blob: 1a575a721d81a45c3b92495154c42b636ed87be8 [file] [log] [blame]
Glenn Kasten99e53b82012-01-19 08:59:58 -08001/*
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9** http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080018//#define LOG_NDEBUG 0
19#define LOG_TAG "AudioTrack"
20
Mark Salyzyn34fb2962014-06-18 16:30:56 -070021#include <inttypes.h>
Glenn Kastenc56f3422014-03-21 17:53:17 -070022#include <math.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080023#include <sys/resource.h>
Andy Hung959b5b82021-09-24 10:46:20 -070024#include <thread>
Mark Salyzyn34fb2962014-06-18 16:30:56 -070025
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -080026#include <android/media/IAudioPolicyService.h>
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -070027#include <android-base/macros.h>
Andy Hung3acde2c2021-11-11 09:18:08 -080028#include <android-base/stringprintf.h>
Andy Hung2b01f002017-07-05 12:01:36 -070029#include <audio_utils/clock.h>
Glenn Kasten9f80dd22012-12-18 15:57:32 -080030#include <audio_utils/primitives.h>
31#include <binder/IPCThreadState.h>
Vlad Popaad0fe922022-06-10 00:43:14 +020032#include <binder/IServiceManager.h>
Glenn Kasten9f80dd22012-12-18 15:57:32 -080033#include <media/AudioTrack.h>
34#include <utils/Log.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080035#include <private/media/AudioTrackShared.h>
Suren Baghdasaryan7435e7d2018-12-19 17:09:28 -080036#include <processgroup/sched_policy.h>
Glenn Kasten1ab85ec2013-05-31 09:18:43 -070037#include <media/IAudioFlinger.h>
Dean Wheatleya70eef72018-01-04 14:23:50 +110038#include <media/AudioParameter.h>
Andy Hungcd044842014-08-07 11:04:34 -070039#include <media/AudioResamplerPublic.h>
Michael Chana94fbb22018-04-24 14:31:19 +100040#include <media/AudioSystem.h>
Ray Essickf27e9872019-12-07 06:28:46 -080041#include <media/MediaMetricsItem.h>
Ray Essicked304702017-12-12 14:00:57 -080042#include <media/TypeConverter.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080043
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +010044#define WAIT_PERIOD_MS 10
45#define WAIT_STREAM_END_TIMEOUT_SEC 120
Vlad Popaad0fe922022-06-10 00:43:14 +020046
Andy Hung53c3b5f2014-12-15 16:42:05 -080047static const int kMaxLoopCountNotifications = 32;
Vlad Popaad0fe922022-06-10 00:43:14 +020048static constexpr char kAudioServiceName[] = "audio";
Glenn Kasten511754b2012-01-11 09:52:19 -080049
Kuowei Lid4adbdb2020-08-13 14:44:25 +080050using ::android::aidl_utils::statusTFromBinderStatus;
Andy Hung3acde2c2021-11-11 09:18:08 -080051using ::android::base::StringPrintf;
Kuowei Lid4adbdb2020-08-13 14:44:25 +080052
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080053namespace android {
Chia-chi Yeh33005a92010-06-16 06:33:13 +080054// ---------------------------------------------------------------------------
55
Ivan Lozano8cf3a072017-08-09 09:01:33 -070056using media::VolumeShaper;
Svet Ganov3e5f14f2021-05-13 22:51:08 +000057using android::content::AttributionSourceState;
Ivan Lozano8cf3a072017-08-09 09:01:33 -070058
Andy Hunga7f03352015-05-31 21:54:49 -070059// TODO: Move to a separate .h
60
Andy Hung4ede21d2014-12-12 15:37:34 -080061template <typename T>
Andy Hunga7f03352015-05-31 21:54:49 -070062static inline const T &min(const T &x, const T &y) {
Andy Hung4ede21d2014-12-12 15:37:34 -080063 return x < y ? x : y;
64}
65
Andy Hunga7f03352015-05-31 21:54:49 -070066template <typename T>
67static inline const T &max(const T &x, const T &y) {
68 return x > y ? x : y;
69}
70
71static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
72{
73 return ((double)frames * 1000000000) / ((double)sampleRate * speed);
74}
75
Andy Hung7f1bc8a2014-09-12 14:43:11 -070076static int64_t convertTimespecToUs(const struct timespec &tv)
77{
Chih-Hung Hsieh50fa06c2018-12-11 13:51:36 -080078 return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -070079}
80
Andy Hungffa36952017-08-17 10:41:51 -070081// TODO move to audio_utils.
82static inline struct timespec convertNsToTimespec(int64_t ns) {
83 struct timespec tv;
84 tv.tv_sec = static_cast<time_t>(ns / NANOS_PER_SECOND);
Andy Hung06a730b2020-04-09 13:28:31 -070085 tv.tv_nsec = static_cast<int64_t>(ns % NANOS_PER_SECOND);
Andy Hungffa36952017-08-17 10:41:51 -070086 return tv;
87}
88
Andy Hung7f1bc8a2014-09-12 14:43:11 -070089// current monotonic time in microseconds.
90static int64_t getNowUs()
91{
92 struct timespec tv;
93 (void) clock_gettime(CLOCK_MONOTONIC, &tv);
94 return convertTimespecToUs(tv);
95}
96
Andy Hung26145642015-04-15 21:56:53 -070097// FIXME: we don't use the pitch setting in the time stretcher (not working);
98// instead we emulate it using our sample rate converter.
99static const bool kFixPitch = true; // enable pitch fix
100static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)
101{
102 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate;
103}
104
105static inline float adjustSpeed(float speed, float pitch)
106{
Ricardo Garcia6c7f0622015-04-30 18:39:16 -0700107 return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
Andy Hung26145642015-04-15 21:56:53 -0700108}
109
110static inline float adjustPitch(float pitch)
111{
112 return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
113}
114
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800115// static
116status_t AudioTrack::getMinFrameCount(
Glenn Kastene33054e2012-11-14 12:54:39 -0800117 size_t* frameCount,
Glenn Kastenfff6d712012-01-12 16:38:12 -0800118 audio_stream_type_t streamType,
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800119 uint32_t sampleRate)
120{
Glenn Kastend65d73c2012-06-22 17:21:07 -0700121 if (frameCount == NULL) {
122 return BAD_VALUE;
123 }
Glenn Kasten04cd0182012-06-25 11:49:27 -0700124
Andy Hung0e48d252015-01-26 11:43:15 -0800125 // FIXME handle in server, like createTrack_l(), possible missing info:
Glenn Kastene0fa4672012-04-24 14:35:14 -0700126 // audio_io_handle_t output
127 // audio_format_t format
128 // audio_channel_mask_t channelMask
Andy Hung0e48d252015-01-26 11:43:15 -0800129 // audio_output_flags_t flags (FAST)
Glenn Kasten3b16c762012-11-14 08:44:39 -0800130 uint32_t afSampleRate;
Glenn Kasten66a04672014-01-08 08:53:44 -0800131 status_t status;
132 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
133 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700134 ALOGE("%s(): Unable to query output sample rate for stream type %d; status %d",
135 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800136 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800137 }
Glenn Kastene33054e2012-11-14 12:54:39 -0800138 size_t afFrameCount;
Glenn Kasten66a04672014-01-08 08:53:44 -0800139 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
140 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700141 ALOGE("%s(): Unable to query output frame count for stream type %d; status %d",
142 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800143 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800144 }
145 uint32_t afLatency;
Glenn Kasten66a04672014-01-08 08:53:44 -0800146 status = AudioSystem::getOutputLatency(&afLatency, streamType);
147 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700148 ALOGE("%s(): Unable to query output latency for stream type %d; status %d",
149 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800150 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800151 }
152
Andy Hung8edb8dc2015-03-26 19:13:55 -0700153 // When called from createTrack, speed is 1.0f (normal speed).
154 // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
Eric Laurent21da6472017-11-09 16:29:26 -0800155 *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
156 sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800157
Andy Hung0e48d252015-01-26 11:43:15 -0800158 // The formula above should always produce a non-zero value under normal circumstances:
159 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
160 // 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 -0800161 if (*frameCount == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700162 ALOGE("%s(): failed for streamType %d, sampleRate %u",
163 __func__, streamType, sampleRate);
Glenn Kasten66a04672014-01-08 08:53:44 -0800164 return BAD_VALUE;
165 }
Andy Hungfb8ede22018-09-12 19:03:24 -0700166 ALOGV("%s(): getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
167 __func__, *frameCount, afFrameCount, afSampleRate, afLatency);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800168 return NO_ERROR;
169}
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800170
Michael Chana94fbb22018-04-24 14:31:19 +1000171// static
172bool AudioTrack::isDirectOutputSupported(const audio_config_base_t& config,
173 const audio_attributes_t& attributes) {
174 ALOGV("%s()", __FUNCTION__);
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800175 const sp<media::IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
Michael Chana94fbb22018-04-24 14:31:19 +1000176 if (aps == 0) return false;
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800177
178 auto result = [&]() -> ConversionResult<bool> {
Mikhail Naganovdbf03642021-08-25 18:15:32 -0700179 media::audio::common::AudioConfigBase configAidl = VALUE_OR_RETURN(
Mikhail Naganov9dec7012021-07-21 10:30:57 -0700180 legacy2aidl_audio_config_base_t_AudioConfigBase(config, false /*isInput*/));
Mikhail Naganov2a1cf612023-05-17 11:48:43 -0700181 media::audio::common::AudioAttributes attributesAidl = VALUE_OR_RETURN(
182 legacy2aidl_audio_attributes_t_AudioAttributes(attributes));
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800183 bool retAidl;
184 RETURN_IF_ERROR(aidl_utils::statusTFromBinderStatus(
185 aps->isDirectOutputSupported(configAidl, attributesAidl, &retAidl)));
186 return retAidl;
187 }();
188 return result.value_or(false);
Michael Chana94fbb22018-04-24 14:31:19 +1000189}
190
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800191// ---------------------------------------------------------------------------
192
Ray Essicked304702017-12-12 14:00:57 -0800193void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
194{
Ray Essick88394302018-01-24 14:52:05 -0800195 // only if we're in a good state...
196 // XXX: shall we gather alternative info if failing?
197 const status_t lstatus = track->initCheck();
198 if (lstatus != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700199 ALOGD("%s(): no metrics gathered, track status=%d", __func__, (int) lstatus);
Ray Essick88394302018-01-24 14:52:05 -0800200 return;
201 }
202
Andy Hungd0979812019-02-21 15:51:44 -0800203#define MM_PREFIX "android.media.audiotrack." // avoid cut-n-paste errors.
Ray Essicked304702017-12-12 14:00:57 -0800204
Andy Hungde602302021-12-07 21:35:49 -0800205 // Do not change this without changing the MediaMetricsService side.
Andy Hungd0979812019-02-21 15:51:44 -0800206 // Java API 28 entries, do not change.
Ray Essickf27e9872019-12-07 06:28:46 -0800207 mMetricsItem->setCString(MM_PREFIX "streamtype", toString(track->streamType()).c_str());
208 mMetricsItem->setCString(MM_PREFIX "type",
Andy Hungd0979812019-02-21 15:51:44 -0800209 toString(track->mAttributes.content_type).c_str());
Ray Essickf27e9872019-12-07 06:28:46 -0800210 mMetricsItem->setCString(MM_PREFIX "usage", toString(track->mAttributes.usage).c_str());
Ray Essicked304702017-12-12 14:00:57 -0800211
Andy Hungd0979812019-02-21 15:51:44 -0800212 // Non-API entries, these can change due to a Java string mistake.
Ray Essickf27e9872019-12-07 06:28:46 -0800213 mMetricsItem->setInt32(MM_PREFIX "sampleRate", (int32_t)track->mSampleRate);
214 mMetricsItem->setInt64(MM_PREFIX "channelMask", (int64_t)track->mChannelMask);
Andy Hungd0979812019-02-21 15:51:44 -0800215 // Non-API entries, these can change.
Ray Essickf27e9872019-12-07 06:28:46 -0800216 mMetricsItem->setInt32(MM_PREFIX "portId", (int32_t)track->mPortId);
217 mMetricsItem->setCString(MM_PREFIX "encoding", toString(track->mFormat).c_str());
218 mMetricsItem->setInt32(MM_PREFIX "frameCount", (int32_t)track->mFrameCount);
219 mMetricsItem->setCString(MM_PREFIX "attributes", toString(track->mAttributes).c_str());
Andy Hung6f451f02021-02-24 21:53:29 -0800220 mMetricsItem->setCString(MM_PREFIX "logSessionId", track->mLogSessionId.c_str());
Andy Hungde602302021-12-07 21:35:49 -0800221 mMetricsItem->setInt32(MM_PREFIX "underrunFrames", (int32_t)track->getUnderrunFrames());
Ray Essicked304702017-12-12 14:00:57 -0800222}
223
Ray Essick88394302018-01-24 14:52:05 -0800224// hand the user a snapshot of the metrics.
Ray Essickf27e9872019-12-07 06:28:46 -0800225status_t AudioTrack::getMetrics(mediametrics::Item * &item)
Ray Essick88394302018-01-24 14:52:05 -0800226{
227 mMediaMetrics.gather(this);
Ray Essickf27e9872019-12-07 06:28:46 -0800228 mediametrics::Item *tmp = mMediaMetrics.dup();
Ray Essick88394302018-01-24 14:52:05 -0800229 if (tmp == nullptr) {
230 return BAD_VALUE;
231 }
232 item = tmp;
233 return NO_ERROR;
234}
Ray Essicked304702017-12-12 14:00:57 -0800235
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000236AudioTrack::AudioTrack(const AttributionSourceState& attributionSource)
Andy Hungdae7f5a2024-04-11 19:01:28 -0700237 : mClientAttributionSource(attributionSource)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800238{
239}
240
241AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800242 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800243 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800244 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700245 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800246 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700247 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400248 const wp<IAudioTrackCallback> & callback,
Atneyaf86d2692021-10-14 14:02:36 -0400249 int32_t notificationFrames,
250 audio_session_t sessionId,
251 transfer_type transferType,
252 const audio_offload_info_t *offloadInfo,
253 const AttributionSourceState& attributionSource,
254 const audio_attributes_t* pAttributes,
255 bool doNotReconnect,
256 float maxRequiredSpeed,
257 audio_port_handle_t selectedDeviceId)
Atneyaf86d2692021-10-14 14:02:36 -0400258{
Andy Hungdae7f5a2024-04-11 19:01:28 -0700259 mSetParams = std::make_unique<SetParams>(
260 streamType, sampleRate, format, channelMask, frameCount, flags, callback,
261 notificationFrames, nullptr /*sharedBuffer*/, false /*threadCanCallJava*/,
262 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
263 doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400264}
265
266namespace {
267 class LegacyCallbackWrapper : public AudioTrack::IAudioTrackCallback {
268 const AudioTrack::legacy_callback_t mCallback;
269 void * const mData;
270 public:
271 LegacyCallbackWrapper(AudioTrack::legacy_callback_t callback, void* user)
272 : mCallback(callback), mData(user) {}
273 size_t onMoreData(const AudioTrack::Buffer & buffer) override {
274 AudioTrack::Buffer copy = buffer;
275 mCallback(AudioTrack::EVENT_MORE_DATA, mData, static_cast<void*>(&copy));
Atneya Nair03079272022-01-18 17:03:14 -0500276 return copy.size();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400277 }
278 void onUnderrun() override {
279 mCallback(AudioTrack::EVENT_UNDERRUN, mData, nullptr);
280 }
281 void onLoopEnd(int32_t loopsRemaining) override {
282 mCallback(AudioTrack::EVENT_LOOP_END, mData, &loopsRemaining);
283 }
284 void onMarker(uint32_t markerPosition) override {
285 mCallback(AudioTrack::EVENT_MARKER, mData, &markerPosition);
286 }
287 void onNewPos(uint32_t newPos) override {
288 mCallback(AudioTrack::EVENT_NEW_POS, mData, &newPos);
289 }
290 void onBufferEnd() override {
291 mCallback(AudioTrack::EVENT_BUFFER_END, mData, nullptr);
292 }
293 void onNewIAudioTrack() override {
294 mCallback(AudioTrack::EVENT_NEW_IAUDIOTRACK, mData, nullptr);
295 }
296 void onStreamEnd() override {
297 mCallback(AudioTrack::EVENT_STREAM_END, mData, nullptr);
298 }
299 size_t onCanWriteMoreData(const AudioTrack::Buffer & buffer) override {
300 AudioTrack::Buffer copy = buffer;
301 mCallback(AudioTrack::EVENT_CAN_WRITE_MORE_DATA, mData, static_cast<void*>(&copy));
Atneya Nair03079272022-01-18 17:03:14 -0500302 return copy.size();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400303 }
304 };
305}
Andreas Huberc8139852012-01-18 10:51:55 -0800306AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800307 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800308 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800309 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700310 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800311 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700312 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400313 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700314 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800315 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000316 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800317 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000318 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700319 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700320 bool doNotReconnect,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700321 float maxRequiredSpeed)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800322{
François Gaffie393f0e02019-04-10 09:09:08 +0200323 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900324
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500325 mSetParams = std::unique_ptr<SetParams>{
326 new SetParams{streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
327 callback, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
328 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
329 doNotReconnect, maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800330}
331
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500332void AudioTrack::onFirstRef() {
333 if (mSetParams) {
334 set(*mSetParams);
335 mSetParams.reset();
336 }
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400337}
338
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800339AudioTrack::~AudioTrack()
340{
Ray Essicked304702017-12-12 14:00:57 -0800341 // pull together the numbers, before we clean up our structures
342 mMediaMetrics.gather(this);
343
Andy Hungb68f5eb2019-12-03 16:49:17 -0800344 mediametrics::LogItem(mMetricsId)
345 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700346 .set(AMEDIAMETRICS_PROP_CALLERNAME,
347 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700348 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700349 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800350 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
351 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
352 .record();
353
Phil Burk7a9577c2021-03-12 20:12:11 +0000354 stopAndJoinCallbacks(); // checks mStatus
355
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800356 if (mStatus == NO_ERROR) {
Marco Nelissenf8880202014-11-14 07:58:25 -0800357 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700358 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700359 mCblkMemory.clear();
360 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800361 IPCThreadState::self()->flushCommands();
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000362 pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
Andy Hungfb8ede22018-09-12 19:03:24 -0700363 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800364 __func__, mPortId,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700365 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
366 AudioSystem::releaseAudioSessionId(mSessionId, clientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800367 }
368}
369
Phil Burk7a9577c2021-03-12 20:12:11 +0000370void AudioTrack::stopAndJoinCallbacks() {
Phil Burk7a9577c2021-03-12 20:12:11 +0000371 // Make sure that callback function exits in the case where
372 // it is looping on buffer full condition in obtainBuffer().
373 // Otherwise the callback thread will never exit.
374 stop();
375 if (mAudioTrackThread != 0) { // not thread safe
Phil Burk7a9577c2021-03-12 20:12:11 +0000376 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
Kuowei Lifdf8e0f2021-11-26 17:38:34 +0800377 mProxy->interrupt();
Phil Burk7a9577c2021-03-12 20:12:11 +0000378 mAudioTrackThread->requestExitAndWait();
379 mAudioTrackThread.clear();
380 }
zhenjun.zhang116df6a2021-12-08 09:17:13 +0800381
382 AutoMutex lock(mLock);
Phil Burk7a9577c2021-03-12 20:12:11 +0000383 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
384 // This may not stop all of these device callbacks!
385 // TODO: Add some sort of protection.
386 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
387 mDeviceCallback.clear();
388 }
389}
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400390status_t AudioTrack::set(
391 audio_stream_type_t streamType,
392 uint32_t sampleRate,
393 audio_format_t format,
394 audio_channel_mask_t channelMask,
395 size_t frameCount,
396 audio_output_flags_t flags,
397 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700398 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800399 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700400 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800401 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000402 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800403 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000404 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700405 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700406 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700407 float maxRequiredSpeed,
408 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800409{
Atneya Nair14aabae2021-11-30 17:36:24 -0500410 LOG_ALWAYS_FATAL_IF(mInitialized, "%s: should not be called twice", __func__);
411 mInitialized = true;
Eric Laurentf32d7812017-11-30 14:44:07 -0800412 status_t status;
413 uint32_t channelCount;
414 pid_t callingPid;
415 pid_t myPid;
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000416 uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
417 pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
Andy Hung3acde2c2021-11-11 09:18:08 -0800418 std::string errorMessage;
Eric Laurent973db022018-11-20 14:54:31 -0800419 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700420 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
wanggang1471f644f2022-07-08 11:10:20 +0800421 "flags %#x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700422 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800423 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000424 sessionId, transferType, attributionSource.uid, attributionSource.pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800425
Phil Burk33ff89b2015-11-30 11:16:01 -0800426 mThreadCanCallJava = threadCanCallJava;
Andy Hungde602302021-12-07 21:35:49 -0800427
428 // These variables are pulled in an error report, so we initialize them early.
jiabin156c6872017-10-06 09:47:15 -0700429 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800430 mSessionId = sessionId;
Andy Hungde602302021-12-07 21:35:49 -0800431 mChannelMask = channelMask;
Andy Hungde602302021-12-07 21:35:49 -0800432 mReqFrameCount = mFrameCount = frameCount;
433 mSampleRate = sampleRate;
434 mOriginalSampleRate = sampleRate;
435 mAttributes = pAttributes != nullptr ? *pAttributes : AUDIO_ATTRIBUTES_INITIALIZER;
436 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Phil Burk33ff89b2015-11-30 11:16:01 -0800437
Eric Laurentd7f33c52022-01-06 13:54:56 +0100438 // update format and flags before storing them in mFormat, mOrigFlags and mFlags
439 if (pAttributes != NULL) {
440 // stream type shouldn't be looked at, this track has audio attributes
441 ALOGV("%s(): Building AudioTrack with attributes:"
442 " usage=%d content=%d flags=0x%x tags=[%s]",
443 __func__,
444 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
445 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
446 }
447
448 // these below should probably come from the audioFlinger too...
449 if (format == AUDIO_FORMAT_DEFAULT) {
450 format = AUDIO_FORMAT_PCM_16_BIT;
451 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
452 flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
453 }
454
455 // force direct flag if format is not linear PCM
456 // or offload was requested
457 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
458 || !audio_is_linear_pcm(format)) {
459 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
460 ? "%s(): Offload request, forcing to Direct Output"
461 : "%s(): Not linear PCM, forcing to Direct Output",
462 __func__);
463 flags = (audio_output_flags_t)
464 // FIXME why can't we allow direct AND fast?
465 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
466 }
467
468 // force direct flag if HW A/V sync requested
469 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
470 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
471 }
472
473 mFormat = format;
474 mOrigFlags = mFlags = flags;
475
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800476 switch (transferType) {
477 case TRANSFER_DEFAULT:
478 if (sharedBuffer != 0) {
479 transferType = TRANSFER_SHARED;
Atneya Nairba809b82022-03-04 18:11:10 -0500480 } else if (callback == nullptr|| threadCanCallJava) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800481 transferType = TRANSFER_SYNC;
482 } else {
483 transferType = TRANSFER_CALLBACK;
484 }
485 break;
486 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700487 case TRANSFER_SYNC_NOTIF_CALLBACK:
Atneya Nairba809b82022-03-04 18:11:10 -0500488 if (callback == nullptr || sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800489 errorMessage = StringPrintf(
490 "%s: Transfer type %s but callback == nullptr || sharedBuffer != 0",
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700491 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800492 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800493 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800494 }
495 break;
496 case TRANSFER_OBTAIN:
497 case TRANSFER_SYNC:
498 if (sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800499 errorMessage = StringPrintf(
500 "%s: Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800501 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800502 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800503 }
504 break;
505 case TRANSFER_SHARED:
506 if (sharedBuffer == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800507 errorMessage = StringPrintf(
508 "%s: Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800509 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800510 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800511 }
512 break;
513 default:
Andy Hung3acde2c2021-11-11 09:18:08 -0800514 errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800515 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800516 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800517 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800518 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800519 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700520 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800521
Andy Hungfb8ede22018-09-12 19:03:24 -0700522 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700523 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800524
Glenn Kasten53cec222013-08-29 09:01:02 -0700525 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700526 if (mAudioTrack != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800527 errorMessage = StringPrintf("%s: Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800528 status = INVALID_OPERATION;
Andy Hung3acde2c2021-11-11 09:18:08 -0800529 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800530 }
531
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800532 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800533 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700534 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800535 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700536 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800537 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800538 errorMessage = StringPrintf("%s: Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800539 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800540 goto error;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700541 }
Andy Hunga2159aa2021-07-20 13:01:52 -0700542 mOriginalStreamType = streamType;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700543 } else {
Andy Hunga2159aa2021-07-20 13:01:52 -0700544 mOriginalStreamType = AUDIO_STREAM_DEFAULT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800545 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800546
547 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700548 if (!audio_is_valid_format(format)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800549 errorMessage = StringPrintf("%s: Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800550 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800551 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800552 }
Eric Laurentc2f1f072009-07-17 12:17:14 -0700553
Glenn Kasten8ba90322013-10-30 11:29:27 -0700554 if (!audio_is_output_channel(channelMask)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800555 errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800556 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800557 goto error;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700558 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800559 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800560 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700561
Dean Wheatleyd883e302023-10-20 06:11:43 +1100562 if (!(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700563 // createTrack will return an error if PCM format is not supported by server,
564 // so no need to check for specific PCM formats here
Dean Wheatleyd883e302023-10-20 06:11:43 +1100565 ALOGW_IF(!audio_has_proportional_frames(format), "%s(): no direct flag for format 0x%x",
566 __func__, format);
Glenn Kastene3aa6592012-12-04 12:22:46 -0800567 }
Dean Wheatleyd883e302023-10-20 06:11:43 +1100568 mFrameSize = audio_bytes_per_frame(channelCount, format);
Glenn Kastene3aa6592012-12-04 12:22:46 -0800569
Eric Laurent0d6db582014-11-12 18:39:44 -0800570 // sampling rate must be specified for direct outputs
Eric Laurentd7f33c52022-01-06 13:54:56 +0100571 if (sampleRate == 0 && (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800572 errorMessage = StringPrintf(
573 "%s: sample rate must be specified for direct outputs", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800574 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800575 goto error;
Eric Laurent0d6db582014-11-12 18:39:44 -0800576 }
Andy Hungff874dc2016-04-11 16:49:09 -0700577 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
578 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800579
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800580 // Make copy of input parameter offloadInfo so that in the future:
581 // (a) createTrack_l doesn't need it as an input parameter
582 // (b) we can support re-creation of offloaded tracks
583 if (offloadInfo != NULL) {
584 mOffloadInfoCopy = *offloadInfo;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800585 } else {
Eric Laurent20b9ef02016-12-05 11:03:16 -0800586 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Ytai Ben-Tsviffa2fd92020-10-20 09:13:53 -0700587 mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
ziyang zhangb3ec8aa2022-05-10 13:28:38 +0800588 mOffloadInfoCopy.format = format;
589 mOffloadInfoCopy.sample_rate = sampleRate;
590 mOffloadInfoCopy.channel_mask = channelMask;
591 mOffloadInfoCopy.stream_type = streamType;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800592 }
593
Glenn Kasten66e46352014-01-16 17:44:23 -0800594 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
595 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800596 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800597 // mFrameCount is initialized in createTrack_l
Glenn Kastenea38ee72016-04-18 11:08:01 -0700598 if (notificationFrames >= 0) {
599 mNotificationFramesReq = notificationFrames;
600 mNotificationsPerBufferReq = 0;
601 } else {
Eric Laurentd7f33c52022-01-06 13:54:56 +0100602 if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800603 errorMessage = StringPrintf(
604 "%s: notificationFrames=%d not permitted for non-fast track",
Andy Hungfb8ede22018-09-12 19:03:24 -0700605 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800606 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800607 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700608 }
609 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700610 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
611 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800612 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800613 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700614 }
615 mNotificationFramesReq = 0;
616 const uint32_t minNotificationsPerBuffer = 1;
617 const uint32_t maxNotificationsPerBuffer = 8;
618 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
619 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
620 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700621 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
622 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700623 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
624 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800625 mNotificationFramesAct = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700626 // TODO b/182392553: refactor or remove
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000627 mClientAttributionSource = AttributionSourceState(attributionSource);
Eric Laurentf32d7812017-11-30 14:44:07 -0800628 callingPid = IPCThreadState::self()->getCallingPid();
629 myPid = getpid();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700630 if (uid == -1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000631 mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700632 IPCThreadState::self()->getCallingUid()));
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800633 }
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700634 if (pid == (pid_t)-1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000635 mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
Marco Nelissend457c972014-02-11 08:47:07 -0800636 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700637 mAuxEffectId = 0;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400638 mCallback = callback;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700639
Atneya Nairba809b82022-03-04 18:11:10 -0500640 if (callback != nullptr) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400641 mAudioTrackThread = sp<AudioTrackThread>::make(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700642 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700643 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700644 }
645
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800646 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100647 {
648 AutoMutex lock(mLock);
649 status = createTrack_l();
650 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700651 if (status != NO_ERROR) {
652 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100653 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
654 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700655 mAudioTrackThread.clear();
656 }
Andy Hung3acde2c2021-11-11 09:18:08 -0800657 // We do not goto error to prevent double-logging errors.
Eric Laurentf32d7812017-11-30 14:44:07 -0800658 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700659 }
660
Andy Hung4ede21d2014-12-12 15:37:34 -0800661 mLoopCount = 0;
662 mLoopStart = 0;
663 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800664 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800665 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700666 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800667 mNewPosition = 0;
668 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700669 mPosition = 0;
670 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700671 mStartNs = 0;
672 mStartFromZeroUs = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700673 AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800674 mSequence = 1;
675 mObservedSequence = mSequence;
676 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700677 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700678 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700679 mTimestampRetrogradePositionReported = false;
680 mTimestampRetrogradeTimeReported = false;
681 mTimestampStallReported = false;
682 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700683 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700684 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800685 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800686 mFramesWritten = 0;
687 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700688 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700689 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800690
Andy Hung3acde2c2021-11-11 09:18:08 -0800691error:
692 if (status != NO_ERROR) {
693 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
694 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
695 }
696 // fall through
Eric Laurentf32d7812017-11-30 14:44:07 -0800697exit:
698 mStatus = status;
699 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800700}
701
Mikhail Naganov55773032020-10-01 15:08:13 -0700702
703status_t AudioTrack::set(
704 audio_stream_type_t streamType,
705 uint32_t sampleRate,
706 audio_format_t format,
707 uint32_t channelMask,
708 size_t frameCount,
709 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400710 legacy_callback_t callback,
Mikhail Naganov55773032020-10-01 15:08:13 -0700711 void* user,
712 int32_t notificationFrames,
713 const sp<IMemory>& sharedBuffer,
714 bool threadCanCallJava,
715 audio_session_t sessionId,
716 transfer_type transferType,
717 const audio_offload_info_t *offloadInfo,
718 uid_t uid,
719 pid_t pid,
720 const audio_attributes_t* pAttributes,
721 bool doNotReconnect,
722 float maxRequiredSpeed,
723 audio_port_handle_t selectedDeviceId)
724{
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000725 AttributionSourceState attributionSource;
726 attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
727 attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
728 attributionSource.token = sp<BBinder>::make();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400729 if (callback) {
730 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
731 } else if (user) {
732 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
733 }
734 return set(streamType, sampleRate, format, static_cast<audio_channel_mask_t>(channelMask),
735 frameCount, flags, mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
736 threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
737 pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Mikhail Naganov55773032020-10-01 15:08:13 -0700738}
739
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800740// -------------------------------------------------------------------------
741
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100742status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800743{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800744 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800745
Andy Hung10fb4be2020-05-27 22:22:22 -0700746 if (mState == STATE_ACTIVE) {
747 return INVALID_OPERATION;
748 }
749
750 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
751
752 // Defer logging here due to OpenSL ES repeated start calls.
753 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
754 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800755 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700756 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800757 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700758 .set(AMEDIAMETRICS_PROP_CALLERNAME,
759 mCallerName.empty()
760 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
761 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800762 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700763 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800764 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
765 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
766 .record(); });
767
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800768
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800769 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800770
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800771 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100772 if (previousState == STATE_PAUSED_STOPPING) {
773 mState = STATE_STOPPING;
774 } else {
775 mState = STATE_ACTIVE;
776 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700777 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700778
779 // save start timestamp
jiabin94ed47c2023-07-27 23:34:20 +0000780 if (isAfTrackOffloadedOrDirect_l()) {
Andy Hung65ffdfc2016-10-10 15:52:11 -0700781 if (getTimestamp_l(mStartTs) != OK) {
782 mStartTs.mPosition = 0;
783 }
784 } else {
785 if (getTimestamp_l(&mStartEts) != OK) {
786 mStartEts.clear();
787 }
788 }
Andy Hungffa36952017-08-17 10:41:51 -0700789 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800790 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
791 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700792 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700793 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700794 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700795 mTimestampRetrogradePositionReported = false;
796 mTimestampRetrogradeTimeReported = false;
797 mTimestampStallReported = false;
798 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700799 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700800
jiabin94ed47c2023-07-27 23:34:20 +0000801 if (!isAfTrackOffloadedOrDirect_l()
Andy Hung65ffdfc2016-10-10 15:52:11 -0700802 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700803 // Server side has consumed something, but is it finished consuming?
804 // It is possible since flush and stop are asynchronous that the server
805 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700806 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800807 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700808 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700809 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
810 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700811 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700812 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
813 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700814 }
Andy Hunge1e98462016-04-12 10:18:51 -0700815 mFramesWritten = 0;
816 mProxy->clearTimestamp(); // need new server push for valid timestamp
817 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700818
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700819 // For offloaded tracks, we don't know if the hardware counters are really zero here,
820 // since the flush is asynchronous and stop may not fully drain.
821 // We save the time when the track is started to later verify whether
822 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700823 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700824
Eric Laurentec9a0322013-08-28 10:23:01 -0700825 // force refresh of remaining frames by processAudioBuffer() as last
826 // write before stop could be partial.
827 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900828
829 // for static track, clear the old flags when starting from stopped state
830 if (mSharedBuffer != 0) {
831 android_atomic_and(
832 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
833 &mCblk->mFlags);
834 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800835 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700836 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700837 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800838
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800839 if (!(flags & CBLK_INVALID)) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -0800840 mAudioTrack->start(&status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800841 if (status == DEAD_OBJECT) {
842 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800843 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800844 }
845 if (flags & CBLK_INVALID) {
846 status = restoreTrack_l("start");
847 }
848
Andy Hung79629f02016-03-24 13:57:40 -0700849 // resume or pause the callback thread as needed.
850 sp<AudioTrackThread> t = mAudioTrackThread;
851 if (status == NO_ERROR) {
852 if (t != 0) {
853 if (previousState == STATE_STOPPING) {
854 mProxy->interrupt();
855 } else {
856 t->resume();
857 }
858 } else {
859 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
860 get_sched_policy(0, &mPreviousSchedulingGroup);
861 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
862 }
Andy Hung39399b62017-04-21 15:07:45 -0700863
864 // Start our local VolumeHandler for restoration purposes.
865 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -0700866 } else {
Eric Laurent973db022018-11-20 14:54:31 -0800867 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800868 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800869 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100870 if (previousState != STATE_STOPPING) {
871 t->pause();
872 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800873 } else {
Glenn Kasten87913512011-06-22 16:15:25 -0700874 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -0700875 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800876 }
877 }
878
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100879 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800880}
881
882void AudioTrack::stop()
883{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800884 const int64_t beginNs = systemTime();
885
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800886 AutoMutex lock(mLock);
Andy Hungb510fcd2024-04-11 19:03:35 -0700887 if (mProxy == nullptr) return; // not successfully initialized.
Andy Hung06a730b2020-04-09 13:28:31 -0700888 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800889 mediametrics::LogItem(mMetricsId)
890 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -0700891 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800892 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -0700893 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
894 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -0700895 .record();
Phil Burka9876702020-04-20 18:16:15 -0700896 });
Andy Hungb68f5eb2019-12-03 16:49:17 -0800897
Eric Laurent973db022018-11-20 14:54:31 -0800898 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700899
Glenn Kasten397edb32013-08-30 15:10:13 -0700900 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800901 return;
902 }
903
Glenn Kasten23a75452014-01-13 10:37:17 -0800904 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100905 mState = STATE_STOPPING;
906 } else {
907 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -0800908 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -0800909 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -0700910 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100911 }
912
Andy Hung1d3556d2018-03-29 16:30:14 -0700913 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800914 mProxy->interrupt();
915 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -0700916
917 // Note: legacy handling - stop does not clear playback marker
918 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -0800919
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800920 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -0800921 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -0800922 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
923 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800924 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100925
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800926 sp<AudioTrackThread> t = mAudioTrackThread;
927 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -0800928 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100929 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -0800930 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -0800931 // causes wake up of the playback thread, that will callback the client for
932 // EVENT_STREAM_END in processAudioBuffer()
933 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100934 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800935 } else {
936 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
937 set_sched_policy(0, mPreviousSchedulingGroup);
938 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800939}
940
941bool AudioTrack::stopped() const
942{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -0800943 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800944 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800945}
946
947void AudioTrack::flush()
948{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800949 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -0700950 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700951 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800952 mediametrics::LogItem(mMetricsId)
953 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -0700954 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800955 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
956 .record(); });
957
Eric Laurent973db022018-11-20 14:54:31 -0800958 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700959
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800960 if (mSharedBuffer != 0) {
961 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -0800962 }
Andy Hung4c5ed302018-05-09 11:16:21 -0700963 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800964 return;
965 }
966 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -0800967}
968
Eric Laurent1703cdf2011-03-07 14:52:59 -0800969void AudioTrack::flush_l()
970{
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800971 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -0700972
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700973 // clear playback marker and periodic update counter
974 mMarkerPosition = 0;
975 mMarkerReached = false;
976 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100977 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -0700978
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800979 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -0700980 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -0800981 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100982 mProxy->interrupt();
983 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800984 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -0800985 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800986}
987
Andy Hung959b5b82021-09-24 10:46:20 -0700988bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
989{
990 using namespace std::chrono_literals;
991
Andy Hungd87a53a2022-01-19 16:56:17 -0800992 // We use atomic access here for state variables - these are used as hints
993 // to ensure we have ramped down audio.
994 const int priorState = mProxy->getState();
995 const uint32_t priorPosition = mProxy->getPosition().unsignedValue();
996
Andy Hung959b5b82021-09-24 10:46:20 -0700997 pause();
998
Andy Hungd87a53a2022-01-19 16:56:17 -0800999 // Only if we were previously active, do we wait to ramp down the audio.
1000 if (priorState != CBLK_STATE_ACTIVE) return true;
1001
Andy Hung959b5b82021-09-24 10:46:20 -07001002 AutoMutex lock(mLock);
1003 // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
1004 if (isOffloadedOrDirect_l()) return true;
1005
1006 // Wait for the track state to be anything besides pausing.
1007 // This ensures that the volume has ramped down.
1008 constexpr auto SLEEP_INTERVAL_MS = 10ms;
Andy Hungd87a53a2022-01-19 16:56:17 -08001009 constexpr auto POSITION_TIMEOUT_MS = 40ms; // don't wait longer than this for position change.
Andy Hung959b5b82021-09-24 10:46:20 -07001010 auto begin = std::chrono::steady_clock::now();
1011 while (true) {
Andy Hungd87a53a2022-01-19 16:56:17 -08001012 // Wait for state and position to change.
1013 // After pause() the server state should be PAUSING, but that may immediately
1014 // convert to PAUSED by prepareTracks before data is read into the mixer.
1015 // Hence we check that the state is not PAUSING and that the server position
1016 // has advanced to be a more reliable estimate that the volume ramp has completed.
Andy Hung959b5b82021-09-24 10:46:20 -07001017 const int state = mProxy->getState();
Andy Hungd87a53a2022-01-19 16:56:17 -08001018 const uint32_t position = mProxy->getPosition().unsignedValue();
Andy Hung959b5b82021-09-24 10:46:20 -07001019
1020 mLock.unlock(); // only local variables accessed until lock.
1021 auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
1022 std::chrono::steady_clock::now() - begin);
Andy Hungd87a53a2022-01-19 16:56:17 -08001023 if (state != CBLK_STATE_PAUSING &&
1024 (elapsed >= POSITION_TIMEOUT_MS || position != priorPosition)) {
1025 ALOGV("%s: success state:%d, position:%u after %lld ms"
1026 " (prior state:%d prior position:%u)",
1027 __func__, state, position, elapsed.count(), priorState, priorPosition);
Andy Hung959b5b82021-09-24 10:46:20 -07001028 return true;
1029 }
1030 std::chrono::milliseconds remaining = timeout - elapsed;
1031 if (remaining.count() <= 0) {
1032 ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
1033 __func__, state, CBLK_STATE_PAUSING, elapsed.count());
1034 return false;
1035 }
1036 // It is conceivable that the track is restored while sleeping;
1037 // as this logic is advisory, we allow that.
1038 std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
1039 mLock.lock();
1040 }
1041}
1042
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001043void AudioTrack::pause()
1044{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001045 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -08001046 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -07001047 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001048 mediametrics::LogItem(mMetricsId)
1049 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -07001050 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001051 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1052 .record(); });
1053
Eric Laurent973db022018-11-20 14:54:31 -08001054 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001055
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001056 if (mState == STATE_ACTIVE) {
1057 mState = STATE_PAUSED;
1058 } else if (mState == STATE_STOPPING) {
1059 mState = STATE_PAUSED_STOPPING;
1060 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001061 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001062 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001063 mProxy->interrupt();
1064 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001065
Marco Nelissen3a90f282014-03-10 11:21:43 -07001066 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001067 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001068 // An offload output can be re-used between two audio tracks having
1069 // the same configuration. A timestamp query for a paused track
1070 // while the other is running would return an incorrect time.
1071 // To fix this, cache the playback position on a pause() and return
1072 // this time when requested until the track is resumed.
1073
1074 // OffloadThread sends HAL pause in its threadLoop. Time saved
1075 // here can be slightly off.
1076
1077 // TODO: check return code for getRenderPosition.
1078
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001079 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001080 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -07001081 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001082 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001083 }
1084 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001085}
1086
Eric Laurentbe916aa2010-06-01 23:49:17 -07001087status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001088{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001089 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1090 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
1091 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001092 return BAD_VALUE;
1093 }
1094
Andy Hungb68f5eb2019-12-03 16:49:17 -08001095 mediametrics::LogItem(mMetricsId)
1096 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
1097 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
1098 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
1099 .record();
1100
Eric Laurent1703cdf2011-03-07 14:52:59 -08001101 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -08001102 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
1103 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001104
Glenn Kastenc56f3422014-03-21 17:53:17 -07001105 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -07001106
Glenn Kasten23a75452014-01-13 10:37:17 -08001107 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -07001108 mAudioTrack->signal();
1109 }
Eric Laurentbe916aa2010-06-01 23:49:17 -07001110 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001111}
1112
Glenn Kastenb1c09932012-02-27 16:21:04 -08001113status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001114{
Glenn Kastenb1c09932012-02-27 16:21:04 -08001115 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001116}
1117
Eric Laurent2beeb502010-07-16 07:43:46 -07001118status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -07001119{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001120 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1121 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001122 return BAD_VALUE;
1123 }
1124
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001125 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001126 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001127 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001128
1129 return NO_ERROR;
1130}
1131
Glenn Kastena5224f32012-01-04 12:41:44 -08001132void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -07001133{
1134 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001135 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001136 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001137}
1138
Glenn Kasten3b16c762012-11-14 08:44:39 -08001139status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001140{
Andy Hung5cbb5782015-03-27 18:39:59 -07001141 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08001142 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -07001143
Andy Hung5cbb5782015-03-27 18:39:59 -07001144 if (rate == mSampleRate) {
1145 return NO_ERROR;
1146 }
jiabinf4de6112018-12-19 12:40:08 -08001147 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
1148 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -08001149 return INVALID_OPERATION;
1150 }
Eric Laurent0d6db582014-11-12 18:39:44 -08001151 if (mOutput == AUDIO_IO_HANDLE_NONE) {
1152 return NO_INIT;
1153 }
Andy Hung5cbb5782015-03-27 18:39:59 -07001154 // NOTE: it is theoretically possible, but highly unlikely, that a device change
1155 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001156 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -08001157 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001158 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001159 }
Andy Hung26145642015-04-15 21:56:53 -07001160 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001161 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001162 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001163 return BAD_VALUE;
1164 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001165 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001166
Glenn Kastene3aa6592012-12-04 12:22:46 -08001167 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001168 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001169
Eric Laurent57326622009-07-07 07:10:45 -07001170 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001171}
1172
Glenn Kastena5224f32012-01-04 12:41:44 -08001173uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001174{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001175 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001176
1177 // sample rate can be updated during playback by the offloaded decoder so we need to
1178 // query the HAL and update if needed.
1179// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001180 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001181 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001182 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001183 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001184 if (status == NO_ERROR) {
1185 mSampleRate = sampleRate;
1186 }
1187 }
1188 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001189 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001190}
1191
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001192uint32_t AudioTrack::getOriginalSampleRate() const
1193{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001194 return mOriginalSampleRate;
1195}
1196
Robert Wu310037a2022-09-06 21:48:18 +00001197uint32_t AudioTrack::getHalSampleRate() const
1198{
1199 return mAfSampleRate;
1200}
1201
1202uint32_t AudioTrack::getHalChannelCount() const
1203{
1204 return mAfChannelCount;
1205}
1206
1207audio_format_t AudioTrack::getHalFormat() const
1208{
1209 return mAfFormat;
1210}
1211
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001212status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1213{
1214 AutoMutex lock(mLock);
1215 return setDualMonoMode_l(mode);
1216}
1217
1218status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1219{
1220 const status_t status = statusTFromBinderStatus(
1221 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1222 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1223 if (status == NO_ERROR) mDualMonoMode = mode;
1224 return status;
1225}
1226
1227status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1228{
1229 AutoMutex lock(mLock);
Mikhail Naganovb1a075b2022-12-18 02:48:14 +00001230 media::audio::common::AudioDualMonoMode mediaMode;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001231 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1232 if (status == NO_ERROR) {
1233 *mode = VALUE_OR_RETURN_STATUS(
1234 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1235 }
1236 return status;
1237}
1238
1239status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1240{
1241 AutoMutex lock(mLock);
1242 return setAudioDescriptionMixLevel_l(leveldB);
1243}
1244
1245status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1246{
1247 const status_t status = statusTFromBinderStatus(
1248 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1249 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1250 return status;
1251}
1252
1253status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1254{
1255 AutoMutex lock(mLock);
1256 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1257}
1258
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001259status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001260{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001261 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001262 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001263 return NO_ERROR;
1264 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001265 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001266 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1267 VALUE_OR_RETURN_STATUS(
1268 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1269 if (status == NO_ERROR) {
1270 mPlaybackRate = playbackRate;
Bertil Ã…kesson081fb742022-07-14 16:39:36 +02001271 } else if (status == INVALID_OPERATION
1272 && playbackRate.mSpeed == 1.0f && mPlaybackRate.mPitch == 1.0f) {
1273 mPlaybackRate = playbackRate;
1274 return NO_ERROR;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001275 }
1276 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001277 }
1278 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1279 return INVALID_OPERATION;
1280 }
Andy Hungff874dc2016-04-11 16:49:09 -07001281
Andy Hungfb8ede22018-09-12 19:03:24 -07001282 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001283 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001284 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001285 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1286 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1287 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001288 AudioPlaybackRate playbackRateTemp = playbackRate;
1289 playbackRateTemp.mSpeed = effectiveSpeed;
1290 playbackRateTemp.mPitch = effectivePitch;
1291
Andy Hungfb8ede22018-09-12 19:03:24 -07001292 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001293 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001294
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001295 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001296 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001297 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001298 return BAD_VALUE;
1299 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001300 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001301 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001302 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001303 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001304 return BAD_VALUE;
1305 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001306
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001307 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001308 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1309 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001310 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001311 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001312 return BAD_VALUE;
1313 }
1314
Dan Austine34eae22015-10-27 16:14:52 -07001315 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001316 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001317 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001318 return BAD_VALUE;
1319 }
1320 mPlaybackRate = playbackRate;
1321 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001322 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001323 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001324
1325 mediametrics::LogItem(mMetricsId)
1326 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1327 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1328 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1329 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1330 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1331 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1332 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1333 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1334 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1335 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1336 .record();
1337
Andy Hung8edb8dc2015-03-26 19:13:55 -07001338 return NO_ERROR;
1339}
1340
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001341const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001342{
1343 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001344 if (isOffloadedOrDirect_l()) {
Mikhail Naganovb1a075b2022-12-18 02:48:14 +00001345 media::audio::common::AudioPlaybackRate playbackRateTemp;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001346 const status_t status = statusTFromBinderStatus(
1347 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1348 if (status == NO_ERROR) { // update local version if changed.
1349 mPlaybackRate =
1350 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1351 }
1352 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001353 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001354}
1355
Phil Burkc0adecb2016-01-08 12:44:11 -08001356ssize_t AudioTrack::getBufferSizeInFrames()
1357{
1358 AutoMutex lock(mLock);
1359 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1360 return NO_INIT;
1361 }
Phil Burka9876702020-04-20 18:16:15 -07001362
Phil Burke8972b02016-03-04 11:29:57 -08001363 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001364}
1365
Andy Hungf2c87b32016-04-07 19:49:29 -07001366status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1367{
1368 if (duration == nullptr) {
1369 return BAD_VALUE;
1370 }
1371 AutoMutex lock(mLock);
1372 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1373 return NO_INIT;
1374 }
1375 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1376 if (bufferSizeInFrames < 0) {
1377 return (status_t)bufferSizeInFrames;
1378 }
1379 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1380 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1381 return NO_ERROR;
1382}
1383
Phil Burkc0adecb2016-01-08 12:44:11 -08001384ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1385{
1386 AutoMutex lock(mLock);
1387 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1388 return NO_INIT;
1389 }
Phil Burka9876702020-04-20 18:16:15 -07001390
1391 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1392 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1393 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001394 android::mediametrics::LogItem(mMetricsId)
1395 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1396 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1397 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1398 .record();
Phil Burka9876702020-04-20 18:16:15 -07001399 }
1400 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001401}
1402
Andy Hung3c7f47a2021-03-16 17:30:09 -07001403ssize_t AudioTrack::getStartThresholdInFrames() const
1404{
1405 AutoMutex lock(mLock);
1406 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1407 return NO_INIT;
1408 }
1409 return (ssize_t) mProxy->getStartThresholdInFrames();
1410}
1411
1412ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1413{
1414 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1415 // contractually we could simply return the current threshold in frames
1416 // to indicate the request was ignored, but we return an error here.
1417 return BAD_VALUE;
1418 }
1419 AutoMutex lock(mLock);
1420 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1421 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1422 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1423 // not have proper validation for the actual set value).
1424 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1425 return NO_INIT;
1426 }
1427 const uint32_t original = mProxy->getStartThresholdInFrames();
1428 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1429 if (original != final) {
1430 android::mediametrics::LogItem(mMetricsId)
1431 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1432 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1433 .record();
1434 if (original > final) {
1435 // restart track if it was disabled by audioflinger due to previous underrun
1436 // and we reduced the number of frames for the threshold.
1437 restartIfDisabled();
1438 }
1439 }
1440 return final;
1441}
1442
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001443status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1444{
Glenn Kastend79072e2016-01-06 08:41:20 -08001445 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001446 return INVALID_OPERATION;
1447 }
1448
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001449 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001450 ;
1451 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1452 loopEnd - loopStart >= MIN_LOOP) {
1453 ;
1454 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001455 return BAD_VALUE;
1456 }
1457
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001458 AutoMutex lock(mLock);
1459 // See setPosition() regarding setting parameters such as loop points or position while active
1460 if (mState == STATE_ACTIVE) {
1461 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001462 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001463 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001464 return NO_ERROR;
1465}
1466
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001467void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1468{
Andy Hung4ede21d2014-12-12 15:37:34 -08001469 // We do not update the periodic notification point.
1470 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1471 mLoopCount = loopCount;
1472 mLoopEnd = loopEnd;
1473 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001474 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001475 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001476
1477 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001478}
1479
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001480status_t AudioTrack::setMarkerPosition(uint32_t marker)
1481{
Atneya Nair14aabae2021-11-30 17:36:24 -05001482 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001483 // The only purpose of setting marker position is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001484 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001485 return INVALID_OPERATION;
1486 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001487
1488 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001489 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001490
Andy Hung3c09c782014-12-29 18:39:32 -08001491 sp<AudioTrackThread> t = mAudioTrackThread;
1492 if (t != 0) {
1493 t->wake();
1494 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001495 return NO_ERROR;
1496}
1497
Glenn Kastena5224f32012-01-04 12:41:44 -08001498status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001499{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001500 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001501 return INVALID_OPERATION;
1502 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001503 if (marker == NULL) {
1504 return BAD_VALUE;
1505 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001506
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001507 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001508 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001509
1510 return NO_ERROR;
1511}
1512
1513status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1514{
Atneya Nair14aabae2021-11-30 17:36:24 -05001515 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001516 // The only purpose of setting position update period is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001517 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001518 return INVALID_OPERATION;
1519 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001520
Glenn Kasten200092b2014-08-15 15:13:30 -07001521 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001522 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001523
Andy Hung3c09c782014-12-29 18:39:32 -08001524 sp<AudioTrackThread> t = mAudioTrackThread;
1525 if (t != 0) {
1526 t->wake();
1527 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001528 return NO_ERROR;
1529}
1530
Glenn Kastena5224f32012-01-04 12:41:44 -08001531status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001532{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001533 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001534 return INVALID_OPERATION;
1535 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001536 if (updatePeriod == NULL) {
1537 return BAD_VALUE;
1538 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001539
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001540 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001541 *updatePeriod = mUpdatePeriod;
1542
1543 return NO_ERROR;
1544}
1545
1546status_t AudioTrack::setPosition(uint32_t position)
1547{
Glenn Kastend79072e2016-01-06 08:41:20 -08001548 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001549 return INVALID_OPERATION;
1550 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001551 if (position > mFrameCount) {
1552 return BAD_VALUE;
1553 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001554
Eric Laurent1703cdf2011-03-07 14:52:59 -08001555 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001556 // Currently we require that the player is inactive before setting parameters such as position
1557 // or loop points. Otherwise, there could be a race condition: the application could read the
1558 // current position, compute a new position or loop parameters, and then set that position or
1559 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1560 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1561 // to specify how it wants to handle such scenarios.
1562 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001563 return INVALID_OPERATION;
1564 }
Andy Hung9b461582014-12-01 17:56:29 -08001565 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001566 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001567 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001568
1569 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001570 return NO_ERROR;
1571}
1572
Glenn Kasten200092b2014-08-15 15:13:30 -07001573status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001574{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001575 if (position == NULL) {
1576 return BAD_VALUE;
1577 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001578
Eric Laurent1703cdf2011-03-07 14:52:59 -08001579 AutoMutex lock(mLock);
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001580 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
1581 if (mState == STATE_STOPPED || mState == STATE_FLUSHED) {
1582 *position = 0;
1583 return NO_ERROR;
1584 }
Andy Hung7a490e72016-03-23 15:58:10 -07001585 // FIXME: offloaded and direct tracks call into the HAL for render positions
1586 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1587 // as we do not know the capability of the HAL for pcm position support and standby.
1588 // There may be some latency differences between the HAL position and the proxy position.
1589 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07001590 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001591 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001592 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001593 *position = mPausedPosition;
1594 return NO_ERROR;
1595 }
1596
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001597 uint32_t dspFrames = 0;
Glenn Kasten142f5192014-03-25 17:44:59 -07001598 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001599 uint32_t halFrames; // actually unused
Andy Hung1f1db832015-06-08 13:26:10 -07001600 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001601 if (AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames) != NO_ERROR) {
1602 *position = 0;
1603 return NO_ERROR;
1604 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001605 }
1606 *position = dspFrames;
1607 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001608 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001609 (void) restoreTrack_l("getPosition");
1610 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1611 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001612 }
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001613 *position = updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001614 }
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001615
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001616 return NO_ERROR;
1617}
1618
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001619status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001620{
Glenn Kastend79072e2016-01-06 08:41:20 -08001621 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001622 return INVALID_OPERATION;
1623 }
1624 if (position == NULL) {
1625 return BAD_VALUE;
1626 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001627
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001628 AutoMutex lock(mLock);
1629 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001630 return NO_ERROR;
1631}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001632
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001633status_t AudioTrack::reload()
1634{
Glenn Kastend79072e2016-01-06 08:41:20 -08001635 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001636 return INVALID_OPERATION;
1637 }
1638
Eric Laurent1703cdf2011-03-07 14:52:59 -08001639 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001640 // See setPosition() regarding setting parameters such as loop points or position while active
1641 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001642 return INVALID_OPERATION;
1643 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001644 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001645 (void) updateAndGetPosition_l();
1646 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001647 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001648#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001649 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001650 // of loop count. Historically we have not restored loop count, start, end,
1651 // but it makes sense if one desires to repeat playing a particular sound.
1652 if (mLoopCount != 0) {
1653 mLoopCountNotified = mLoopCount;
1654 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1655 }
1656#endif
Andy Hung9b461582014-12-01 17:56:29 -08001657 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001658 return NO_ERROR;
1659}
1660
Glenn Kasten38e905b2014-01-13 10:21:48 -08001661audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001662{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001663 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001664 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001665}
1666
Paul McLeanaa981192015-03-21 09:55:15 -07001667status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
Kuowei Li72c8b062023-08-31 13:38:32 +08001668 status_t result = NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001669 AutoMutex lock(mLock);
Kuowei Li72c8b062023-08-31 13:38:32 +08001670 ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
1671 __func__, mPortId, deviceId, mSelectedDeviceId);
Paul McLeanaa981192015-03-21 09:55:15 -07001672 if (mSelectedDeviceId != deviceId) {
1673 mSelectedDeviceId = deviceId;
Dorin Drimusefc130c2024-01-12 16:51:56 +00001674 if (mStatus == NO_ERROR) {
Kuowei Li72c8b062023-08-31 13:38:32 +08001675 if (isOffloadedOrDirect_l()) {
gmanam7b69bd42024-04-26 14:46:10 +05301676 if (isPlaying_l()) {
Kuowei Li72c8b062023-08-31 13:38:32 +08001677 ALOGW("%s(%d). Offloaded or Direct track is not STOPPED or FLUSHED. "
1678 "State: %s.",
1679 __func__, mPortId, stateToString(mState));
1680 result = INVALID_OPERATION;
gmanam7b69bd42024-04-26 14:46:10 +05301681 } else {
1682 ALOGD("%s(%d): creating a new AudioTrack", __func__, mPortId);
1683 result = restoreTrack_l("setOutputDevice", true /* forceRestore */);
Dorin Drimusefc130c2024-01-12 16:51:56 +00001684 }
Eric Laurent72af8012023-03-15 17:36:22 +01001685 } else {
Kuowei Li72c8b062023-08-31 13:38:32 +08001686 // allow track invalidation when track is not playing to propagate
1687 // the updated mSelectedDeviceId
1688 if (isPlaying_l()) {
1689 if (mSelectedDeviceId != mRoutedDeviceId) {
1690 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
1691 mProxy->interrupt();
1692 }
1693 } else {
1694 // if the track is idle, try to restore now and
1695 // defer to next start if not possible
1696 if (restoreTrack_l("setOutputDevice") != OK) {
1697 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
1698 }
Eric Laurent72af8012023-03-15 17:36:22 +01001699 }
1700 }
Eric Laurentfb00fc72017-05-25 18:17:12 -07001701 }
Paul McLeanaa981192015-03-21 09:55:15 -07001702 }
Kuowei Li72c8b062023-08-31 13:38:32 +08001703 return result;
Paul McLeanaa981192015-03-21 09:55:15 -07001704}
1705
1706audio_port_handle_t AudioTrack::getOutputDevice() {
1707 AutoMutex lock(mLock);
1708 return mSelectedDeviceId;
1709}
1710
Eric Laurentad2e7b92017-09-14 20:06:42 -07001711// must be called with mLock held
1712void AudioTrack::updateRoutedDeviceId_l()
1713{
1714 // if the track is inactive, do not update actual device as the output stream maybe routed
1715 // to a device not relevant to this client because of other active use cases.
1716 if (mState != STATE_ACTIVE) {
1717 return;
1718 }
1719 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1720 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1721 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1722 mRoutedDeviceId = deviceId;
1723 }
1724 }
1725}
1726
Eric Laurent296fb132015-05-01 11:38:42 -07001727audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1728 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001729 updateRoutedDeviceId_l();
1730 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001731}
1732
Eric Laurentbe916aa2010-06-01 23:49:17 -07001733status_t AudioTrack::attachAuxEffect(int effectId)
1734{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001735 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001736 status_t status;
1737 mAudioTrack->attachAuxEffect(effectId, &status);
Eric Laurent2beeb502010-07-16 07:43:46 -07001738 if (status == NO_ERROR) {
1739 mAuxEffectId = effectId;
1740 }
1741 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001742}
1743
Eric Laurente83b55d2014-11-14 10:06:21 -08001744audio_stream_type_t AudioTrack::streamType() const
1745{
Eric Laurente83b55d2014-11-14 10:06:21 -08001746 return mStreamType;
1747}
1748
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001749uint32_t AudioTrack::latency()
1750{
1751 AutoMutex lock(mLock);
1752 updateLatency_l();
1753 return mLatency;
1754}
1755
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001756// -------------------------------------------------------------------------
1757
Eric Laurent1703cdf2011-03-07 14:52:59 -08001758// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001759void AudioTrack::updateLatency_l()
1760{
1761 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1762 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001763 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001764 } else {
1765 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001766 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001767 }
1768}
1769
Phil Burkadbb75a2017-06-16 12:19:42 -07001770// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1771#define MEDIA_CASE_ENUM(name) case name: return #name
1772const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1773 switch (transferType) {
1774 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1775 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1776 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1777 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1778 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001779 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001780 default:
1781 return "UNRECOGNIZED";
1782 }
1783}
1784
Glenn Kasten200092b2014-08-15 15:13:30 -07001785status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001786{
Eric Laurentf32d7812017-11-30 14:44:07 -08001787 status_t status;
1788 bool callbackAdded = false;
Andy Hung3acde2c2021-11-11 09:18:08 -08001789 std::string errorMessage;
Eric Laurentf32d7812017-11-30 14:44:07 -08001790
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001791 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1792 if (audioFlinger == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001793 errorMessage = StringPrintf("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001794 __func__, mPortId);
Andy Hung3acde2c2021-11-11 09:18:08 -08001795 status = DEAD_OBJECT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001796 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001797 }
1798
Eric Laurent21da6472017-11-09 16:29:26 -08001799 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001800 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1801 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001802 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001803 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001804 // either of these use cases:
1805 // use case 1: shared buffer
1806 bool sharedBuffer = mSharedBuffer != 0;
1807 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001808 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001809 (mTransfer == TRANSFER_CALLBACK) ||
1810 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001811 (mTransfer == TRANSFER_OBTAIN) ||
1812 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001813 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1814 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001815
Eric Laurent21da6472017-11-09 16:29:26 -08001816 bool fastAllowed = sharedBuffer || transferAllowed;
1817 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001818 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1819 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001820 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001821 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001822 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1823 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001824 }
1825
Eric Laurent21da6472017-11-09 16:29:26 -08001826 IAudioFlinger::CreateTrackInput input;
Andy Hunga2159aa2021-07-20 13:01:52 -07001827 if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
1828 // Legacy: This is based on original parameters even if the track is recreated.
1829 input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001830 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001831 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001832 }
Eric Laurent21da6472017-11-09 16:29:26 -08001833 input.config = AUDIO_CONFIG_INITIALIZER;
1834 input.config.sample_rate = mSampleRate;
1835 input.config.channel_mask = mChannelMask;
1836 input.config.format = mFormat;
1837 input.config.offload_info = mOffloadInfoCopy;
Svet Ganov3e5f14f2021-05-13 22:51:08 +00001838 input.clientInfo.attributionSource = mClientAttributionSource;
Eric Laurent21da6472017-11-09 16:29:26 -08001839 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001840 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001841 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1842 // application-level code follows all non-blocking design rules, the language runtime
1843 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001844 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001845 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001846 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001847 }
Eric Laurent21da6472017-11-09 16:29:26 -08001848 input.sharedBuffer = mSharedBuffer;
1849 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1850 input.speed = 1.0;
1851 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1852 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1853 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1854 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1855 }
1856 input.flags = mFlags;
1857 input.frameCount = mReqFrameCount;
1858 input.notificationFrameCount = mNotificationFramesReq;
1859 input.selectedDeviceId = mSelectedDeviceId;
1860 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001861 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001862
Ytai Ben-Tsvi4dfeb622020-11-02 12:47:30 -08001863 media::CreateTrackResponse response;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001864 status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
Ytai Ben-Tsvi357e26a2021-01-05 13:21:19 -08001865
1866 IAudioFlinger::CreateTrackOutput output{};
1867 if (status == NO_ERROR) {
1868 output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
1869 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001870
Eric Laurent21da6472017-11-09 16:29:26 -08001871 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001872 errorMessage = StringPrintf(
1873 "%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001874 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001875 if (status == NO_ERROR) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001876 status = INVALID_OPERATION; // device not ready
Eric Laurentf32d7812017-11-30 14:44:07 -08001877 }
1878 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001879 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001880 ALOG_ASSERT(output.audioTrack != 0);
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001881
Eric Laurent21da6472017-11-09 16:29:26 -08001882 mFrameCount = output.frameCount;
1883 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1884 mRoutedDeviceId = output.selectedDeviceId;
1885 mSessionId = output.sessionId;
Andy Hunga2159aa2021-07-20 13:01:52 -07001886 mStreamType = output.streamType;
Eric Laurent21da6472017-11-09 16:29:26 -08001887
1888 mSampleRate = output.sampleRate;
1889 if (mOriginalSampleRate == 0) {
1890 mOriginalSampleRate = mSampleRate;
1891 }
1892
1893 mAfFrameCount = output.afFrameCount;
1894 mAfSampleRate = output.afSampleRate;
Robert Wu310037a2022-09-06 21:48:18 +00001895 mAfChannelCount = audio_channel_count_from_out_mask(output.afChannelMask);
1896 mAfFormat = output.afFormat;
Eric Laurent21da6472017-11-09 16:29:26 -08001897 mAfLatency = output.afLatencyMs;
jiabin94ed47c2023-07-27 23:34:20 +00001898 mAfTrackFlags = output.afTrackFlags;
Eric Laurent21da6472017-11-09 16:29:26 -08001899
1900 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1901
Glenn Kasten38e905b2014-01-13 10:21:48 -08001902 // AudioFlinger now owns the reference to the I/O handle,
1903 // so we are no longer responsible for releasing it.
1904
Glenn Kasten7fd04222016-02-02 12:38:16 -08001905 // FIXME compare to AudioRecord
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001906 std::optional<media::SharedFileRegion> sfr;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001907 output.audioTrack->getCblk(&sfr);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001908 sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001909 if (iMem == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001910 errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId);
1911 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001912 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001913 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001914 // TODO: Using unsecurePointer() has some associated security pitfalls
1915 // (see declaration for details).
1916 // Either document why it is safe in this case or address the
1917 // issue (e.g. by copying).
1918 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08001919 if (iMemPointer == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001920 errorMessage = StringPrintf(
1921 "%s(%d): Could not get control block pointer", __func__, mPortId);
1922 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001923 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08001924 }
Glenn Kasten53cec222013-08-29 09:01:02 -07001925 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001926 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08001927 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001928 mDeathNotifier.clear();
1929 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001930 mAudioTrack = output.audioTrack;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001931 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07001932 IPCThreadState::self()->flushCommands();
1933
Glenn Kasten0cde0762014-01-16 15:06:36 -08001934 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001935 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08001936
Glenn Kastena07f17c2013-04-23 12:39:37 -07001937 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08001938 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08001939 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001940 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001941 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08001942 if (!mThreadCanCallJava) {
1943 mAwaitBoost = true;
1944 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001945 } else {
Robert Wuc121cd12021-08-13 17:51:40 +00001946 ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001947 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001948 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001949 }
Eric Laurent21da6472017-11-09 16:29:26 -08001950 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001951
Eric Laurentad2e7b92017-09-14 20:06:42 -07001952 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07001953 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07001954 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07001955 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001956 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07001957 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001958 callbackAdded = true;
1959 }
1960
Eric Laurent09f1ed22019-04-24 17:45:17 -07001961 mPortId = output.portId;
Vlad Popaad0fe922022-06-10 00:43:14 +02001962 // notify the upper layers about the new portId
1963 triggerPortIdUpdate_l();
1964
Glenn Kasten38e905b2014-01-13 10:21:48 -08001965 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08001966 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001967 mRefreshRemaining = true;
1968
1969 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
1970 // is the value of pointer() for the shared buffer, otherwise buffers points
1971 // immediately after the control block. This address is for the mapping within client
1972 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
1973 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08001974 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07001975 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001976 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001977 // TODO: Using unsecurePointer() has some associated security pitfalls
1978 // (see declaration for details).
1979 // Either document why it is safe in this case or address the
1980 // issue (e.g. by copying).
1981 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07001982 if (buffers == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001983 errorMessage = StringPrintf(
1984 "%s(%d): Could not get buffer pointer", __func__, mPortId);
1985 ALOGE("%s", errorMessage.c_str());
1986 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001987 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07001988 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001989 }
1990
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001991 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08001992
Glenn Kasten093000f2012-05-03 09:35:36 -07001993 // If IAudioTrack is re-created, don't let the requested frameCount
1994 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08001995 if (mFrameCount > mReqFrameCount) {
1996 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07001997 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001998
Andy Hungd7bd69e2015-07-24 07:52:41 -07001999 // reset server position to 0 as we have new cblk.
2000 mServer = 0;
2001
Glenn Kastene3aa6592012-12-04 12:22:46 -08002002 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08002003 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002004 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08002005 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002006 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08002007 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002008 mProxy = mStaticProxy;
2009 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002010
2011 mProxy->setVolumeLR(gain_minifloat_pack(
2012 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2013 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2014
Glenn Kastene3aa6592012-12-04 12:22:46 -08002015 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002016 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2017 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2018 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002019 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002020
2021 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2022 playbackRateTemp.mSpeed = effectiveSpeed;
2023 playbackRateTemp.mPitch = effectivePitch;
2024 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002025 mProxy->setMinimum(mNotificationFramesAct);
2026
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002027 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2028 setDualMonoMode_l(mDualMonoMode);
2029 }
2030 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2031 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2032 }
2033
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002034 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002035 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002036
Andy Hungb68f5eb2019-12-03 16:49:17 -08002037 // This is the first log sent from the AudioTrack client.
2038 // The creation of the audio track by AudioFlinger (in the code above)
2039 // is the first log of the AudioTrack and must be present before
2040 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002041
Andy Hungb68f5eb2019-12-03 16:49:17 -08002042 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2043 mediametrics::LogItem(mMetricsId)
2044 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2045 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002046 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2047 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002048 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002049 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002050 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002051 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002052 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2053 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2054 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2055 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2056 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2057 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2058 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2059 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2060 // the following are NOT immutable
2061 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2062 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2063 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Andy Hung73dc2f92021-12-07 21:50:04 -08002064 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)NO_ERROR)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002065 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2066 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2067 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2068 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2069 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2070 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2071 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2072 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2073 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2074 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2075 .record();
2076
2077 // mSendLevel
2078 // mReqFrameCount?
2079 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2080 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2081
Glenn Kasten38e905b2014-01-13 10:21:48 -08002082 }
2083
Eric Laurentf32d7812017-11-30 14:44:07 -08002084exit:
Andy Hung3acde2c2021-11-11 09:18:08 -08002085 if (status != NO_ERROR) {
2086 if (callbackAdded) {
2087 // note: mOutput is always valid is callbackAdded is true
2088 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
2089 }
2090 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
2091 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
Eric Laurentad2e7b92017-09-14 20:06:42 -07002092 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002093 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002094
2095 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002096 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002097}
2098
Andy Hung3acde2c2021-11-11 09:18:08 -08002099void AudioTrack::reportError(status_t status, const char *event, const char *message) const
2100{
2101 if (status == NO_ERROR) return;
2102 // We report error on the native side because some callers do not come
2103 // from Java.
Andy Hungde602302021-12-07 21:35:49 -08002104 // Ensure these variables are initialized in set().
2105 mediametrics::LogItem(AMEDIAMETRICS_KEY_AUDIO_TRACK_ERROR)
Andy Hung3acde2c2021-11-11 09:18:08 -08002106 .set(AMEDIAMETRICS_PROP_EVENT, event)
Andy Hung73dc2f92021-12-07 21:50:04 -08002107 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
2108 .set(AMEDIAMETRICS_PROP_STATUSMESSAGE, message)
Andy Hung3acde2c2021-11-11 09:18:08 -08002109 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
2110 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
2111 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2112 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2113 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2114 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2115 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
Andy Hung3acde2c2021-11-11 09:18:08 -08002116 // the following are NOT immutable
Andy Hungde602302021-12-07 21:35:49 -08002117 // frame count is initially the requested frame count, but may be adjusted
2118 // by AudioFlinger after creation.
2119 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
Andy Hung3acde2c2021-11-11 09:18:08 -08002120 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2121 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2122 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2123 .record();
2124}
2125
Glenn Kastenb46f3942015-03-09 12:00:30 -07002126status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002127{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002128 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002129 if (nonContig != NULL) {
2130 *nonContig = 0;
2131 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002132 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002133 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002134 if (mTransfer != TRANSFER_OBTAIN) {
2135 audioBuffer->frameCount = 0;
Atneya Nair03079272022-01-18 17:03:14 -05002136 audioBuffer->mSize = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002137 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002138 if (nonContig != NULL) {
2139 *nonContig = 0;
2140 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002141 return INVALID_OPERATION;
2142 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002143
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002144 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002145 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002146 if (waitCount == -1) {
2147 requested = &ClientProxy::kForever;
2148 } else if (waitCount == 0) {
2149 requested = &ClientProxy::kNonBlocking;
2150 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002151 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002152 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002153 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002154 requested = &timeout;
2155 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002156 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002157 requested = NULL;
2158 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002159 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002160}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002161
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002162status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2163 struct timespec *elapsed, size_t *nonContig)
2164{
2165 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2166 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002167
2168 Proxy::Buffer buffer;
2169 status_t status = NO_ERROR;
2170
2171 static const int32_t kMaxTries = 5;
2172 int32_t tryCounter = kMaxTries;
2173
2174 do {
2175 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2176 // keep them from going away if another thread re-creates the track during obtainBuffer()
2177 sp<AudioTrackClientProxy> proxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002178
2179 { // start of lock scope
2180 AutoMutex lock(mLock);
2181
Glenn Kasten305996c2020-01-27 08:03:37 -08002182 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002183 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2184 if (status == DEAD_OBJECT) {
2185 // re-create track, unless someone else has already done so
2186 if (newSequence == oldSequence) {
2187 status = restoreTrack_l("obtainBuffer");
2188 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002189 buffer.mFrameCount = 0;
2190 buffer.mRaw = NULL;
2191 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002192 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002193 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002194 }
2195 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002196 oldSequence = newSequence;
2197
Eric Laurent4d231dc2016-03-11 18:38:23 -08002198 if (status == NOT_ENOUGH_DATA) {
2199 restartIfDisabled();
2200 }
2201
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002202 // Keep the extra references
jiabind42567c2023-03-23 22:01:16 +00002203 mProxyObtainBufferRef = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002204 proxy = mProxy;
jiabind42567c2023-03-23 22:01:16 +00002205 mCblkMemoryObtainBufferRef = mCblkMemory;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002206
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002207 if (mState == STATE_STOPPING) {
2208 status = -EINTR;
2209 buffer.mFrameCount = 0;
2210 buffer.mRaw = NULL;
2211 buffer.mNonContig = 0;
2212 break;
2213 }
2214
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002215 // Non-blocking if track is stopped or paused
2216 if (mState != STATE_ACTIVE) {
2217 requested = &ClientProxy::kNonBlocking;
2218 }
2219
2220 } // end of lock scope
2221
2222 buffer.mFrameCount = audioBuffer->frameCount;
2223 // FIXME starts the requested timeout and elapsed over from scratch
2224 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002225 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002226
2227 audioBuffer->frameCount = buffer.mFrameCount;
Atneya Nair03079272022-01-18 17:03:14 -05002228 audioBuffer->mSize = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002229 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002230 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002231 if (nonContig != NULL) {
2232 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002233 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002234 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002235}
2236
Glenn Kasten54a8a452015-03-09 12:03:00 -07002237void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002238{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002239 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002240 if (mTransfer == TRANSFER_SHARED) {
2241 return;
2242 }
2243
Atneya Nair03079272022-01-18 17:03:14 -05002244 size_t stepCount = audioBuffer->mSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002245 if (stepCount == 0) {
2246 return;
2247 }
2248
2249 Proxy::Buffer buffer;
2250 buffer.mFrameCount = stepCount;
2251 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002252
jiabind42567c2023-03-23 22:01:16 +00002253 sp<IMemory> tempMemory;
2254 sp<AudioTrackClientProxy> tempProxy;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002255 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002256 if (audioBuffer->sequence != mSequence) {
2257 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2258 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2259 __func__, audioBuffer->sequence, mSequence);
2260 return;
2261 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002262 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002263 mInUnderrun = false;
jiabind42567c2023-03-23 22:01:16 +00002264 mProxyObtainBufferRef->releaseBuffer(&buffer);
2265 // The extra reference of shared memory and proxy from `obtainBuffer` is not used after
2266 // calling `releaseBuffer`. Move the extra reference to a temp strong pointer so that it
2267 // will be cleared outside `releaseBuffer`.
2268 tempMemory = std::move(mCblkMemoryObtainBufferRef);
2269 tempProxy = std::move(mProxyObtainBufferRef);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002270
2271 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002272 restartIfDisabled();
2273}
2274
2275void AudioTrack::restartIfDisabled()
2276{
2277 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2278 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002279 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002280 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002281 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002282 status_t status;
2283 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002284 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002285}
2286
2287// -------------------------------------------------------------------------
2288
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002289ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002290{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002291 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002292 return INVALID_OPERATION;
2293 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002294
Eric Laurentab5cdba2014-06-09 17:22:27 -07002295 if (isDirect()) {
2296 AutoMutex lock(mLock);
2297 int32_t flags = android_atomic_and(
2298 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2299 &mCblk->mFlags);
2300 if (flags & CBLK_INVALID) {
2301 return DEAD_OBJECT;
2302 }
2303 }
2304
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002305 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002306 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002307 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002308 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002309 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002310 return BAD_VALUE;
2311 }
2312
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002313 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002314 Buffer audioBuffer;
2315
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002316 while (userSize >= mFrameSize) {
2317 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002318
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002319 status_t err = obtainBuffer(&audioBuffer,
2320 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002321 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002322 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002323 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002324 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002325 if (err == TIMED_OUT || err == -EINTR) {
2326 err = WOULD_BLOCK;
2327 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002328 return ssize_t(err);
2329 }
2330
Atneya Nair03079272022-01-18 17:03:14 -05002331 size_t toWrite = audioBuffer.size();
2332 memcpy(audioBuffer.raw, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002333 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002334 userSize -= toWrite;
2335 written += toWrite;
2336
2337 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002338 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002339
Andy Hungea2b9c02016-02-12 17:06:53 -08002340 if (written > 0) {
2341 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002342
2343 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2344 const sp<AudioTrackThread> t = mAudioTrackThread;
2345 if (t != 0) {
2346 // causes wake up of the playback thread, that will callback the client for
2347 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2348 t->wake();
2349 }
2350 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002351 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002352
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002353 return written;
2354}
2355
2356// -------------------------------------------------------------------------
2357
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002358nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002359{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002360 // Currently the AudioTrack thread is not created if there are no callbacks.
2361 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002362 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002363 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002364 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002365 sp<IAudioTrackCallback> callback = mCallback.promote();
2366 if (!callback) {
2367 mCallback = nullptr;
Atneya Naire260f5a2022-05-03 17:02:20 -04002368 mLock.unlock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002369 return NS_NEVER;
2370 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002371 if (mAwaitBoost) {
2372 mAwaitBoost = false;
2373 mLock.unlock();
2374 static const int32_t kMaxTries = 5;
2375 int32_t tryCounter = kMaxTries;
2376 uint32_t pollUs = 10000;
2377 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002378 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002379 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2380 break;
2381 }
2382 usleep(pollUs);
2383 pollUs <<= 1;
2384 } while (tryCounter-- > 0);
2385 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002386 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002387 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002388 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002389 // Run again immediately
2390 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002391 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002392
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002393 // Can only reference mCblk while locked
2394 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002395 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002396
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002397 // Check for track invalidation
2398 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002399 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2400 // AudioSystem cache. We should not exit here but after calling the callback so
2401 // that the upper layers can recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002402 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002403 status_t status __unused = restoreTrack_l("processAudioBuffer");
2404 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002405 // after restoration, continue below to make sure that the loop and buffer events
2406 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002407 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002408 }
2409
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002410 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002411 bool active = mState == STATE_ACTIVE;
2412
2413 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2414 bool newUnderrun = false;
2415 if (flags & CBLK_UNDERRUN) {
2416#if 0
2417 // Currently in shared buffer mode, when the server reaches the end of buffer,
2418 // the track stays active in continuous underrun state. It's up to the application
2419 // to pause or stop the track, or set the position to a new offset within buffer.
2420 // This was some experimental code to auto-pause on underrun. Keeping it here
2421 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2422 if (mTransfer == TRANSFER_SHARED) {
2423 mState = STATE_PAUSED;
2424 active = false;
2425 }
2426#endif
2427 if (!mInUnderrun) {
2428 mInUnderrun = true;
2429 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002430 }
2431 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002432
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002433 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002434 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002435
2436 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002437 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002438 Modulo<uint32_t> markerPosition(mMarkerPosition);
2439 // uses 32 bit wraparound for comparison with position.
2440 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002441 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002442 }
2443
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002444 // Determine number of new position callback(s) that will be needed, while locked
2445 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002446 Modulo<uint32_t> newPosition(mNewPosition);
2447 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002448 // FIXME fails for wraparound, need 64 bits
2449 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002450 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002451 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002452 }
2453
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002454 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002455 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002456 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002457 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002458 if (mRefreshRemaining) {
2459 mRefreshRemaining = false;
2460 mRemainingFrames = notificationFrames;
2461 mRetryOnPartialBuffer = false;
2462 }
2463 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002464 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002465 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002466
Andy Hung53c3b5f2014-12-15 16:42:05 -08002467 // Determine the number of new loop callback(s) that will be needed, while locked.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002468 uint32_t loopCountNotifications = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -08002469 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2470
2471 if (mLoopCount > 0) {
2472 int loopCount;
2473 size_t bufferPosition;
2474 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2475 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2476 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2477 mLoopCountNotified = loopCount; // discard any excess notifications
2478 } else if (mLoopCount < 0) {
2479 // FIXME: We're not accurate with notification count and position with infinite looping
2480 // since loopCount from server side will always return -1 (we could decrement it).
2481 size_t bufferPosition = mStaticProxy->getBufferPosition();
2482 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2483 loopPeriod = mLoopEnd - bufferPosition;
2484 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2485 size_t bufferPosition = mStaticProxy->getBufferPosition();
2486 loopPeriod = mFrameCount - bufferPosition;
2487 }
2488
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002489 // These fields don't need to be cached, because they are assigned only by set():
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002490 // mTransfer, mCallback, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002491 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2492
2493 mLock.unlock();
2494
Andy Hunga7f03352015-05-31 21:54:49 -07002495 // get anchor time to account for callbacks.
2496 const nsecs_t timeBeforeCallbacks = systemTime();
2497
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002498 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002499 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2500 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2501 // (and make sure we don't callback for more data while we're stopping).
2502 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002503 struct timespec timeout;
2504 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2505 timeout.tv_nsec = 0;
2506
Andy Hungeb0732d2023-03-29 20:31:47 -07002507 // Use timestamp progress to safeguard we don't falsely time out.
2508 AudioTimestamp timestamp{};
2509 const bool isTimestampValid = getTimestamp(timestamp) == OK;
2510 const auto frameCount = isTimestampValid ? timestamp.mPosition : 0;
2511
Glenn Kasten96f04882013-09-20 09:28:56 -07002512 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002513 switch (status) {
Andy Hungeb0732d2023-03-29 20:31:47 -07002514 case TIMED_OUT:
2515 if (isTimestampValid
2516 && getTimestamp(timestamp) == OK && frameCount != timestamp.mPosition) {
2517 ALOGD("%s: waitStreamEndDone retrying", __func__);
2518 break; // we retry again (and recheck possible state change).
2519 }
2520 [[fallthrough]];
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002521 case NO_ERROR:
2522 case DEAD_OBJECT:
Andy Hung39609a02015-09-03 16:38:38 -07002523 if (status != DEAD_OBJECT) {
2524 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2525 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002526 callback->onStreamEnd();
Andy Hung39609a02015-09-03 16:38:38 -07002527 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002528 {
2529 AutoMutex lock(mLock);
2530 // The previously assigned value of waitStreamEnd is no longer valid,
2531 // since the mutex has been unlocked and either the callback handler
2532 // or another thread could have re-started the AudioTrack during that time.
2533 waitStreamEnd = mState == STATE_STOPPING;
2534 if (waitStreamEnd) {
2535 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002536 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002537 }
2538 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002539 if (waitStreamEnd && status != DEAD_OBJECT) {
Andy Hungeb0732d2023-03-29 20:31:47 -07002540 ALOGV("%s: waitStreamEndDone complete", __func__);
Glenn Kasten96f04882013-09-20 09:28:56 -07002541 return NS_INACTIVE;
2542 }
2543 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002544 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002545 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002546 }
2547
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002548 // perform callbacks while unlocked
2549 if (newUnderrun) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002550 callback->onUnderrun();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002551 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002552 while (loopCountNotifications > 0) {
Andy Hung53c3b5f2014-12-15 16:42:05 -08002553 --loopCountNotifications;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002554 callback->onLoopEnd(mLoopCount > 0 ? loopCountNotifications + mLoopCountNotified : -1);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002555 }
2556 if (flags & CBLK_BUFFER_END) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002557 callback->onBufferEnd();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002558 }
2559 if (markerReached) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002560 callback->onMarker(markerPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002561 }
2562 while (newPosCount > 0) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002563 callback->onNewPos(newPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002564 newPosition += updatePeriod;
2565 newPosCount--;
2566 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002567
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002568 if (mObservedSequence != sequence) {
2569 mObservedSequence = sequence;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002570 callback->onNewIAudioTrack();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002571 // for offloaded tracks, just wait for the upper layers to recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002572 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002573 return NS_INACTIVE;
2574 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002575 }
2576
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002577 // if inactive, then don't run me again until re-started
2578 if (!active) {
2579 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002580 }
2581
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002582 // Compute the estimated time until the next timed event (position, markers, loops)
2583 // FIXME only for non-compressed audio
2584 uint32_t minFrames = ~0;
2585 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002586 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002587 }
2588 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002589 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002590 minFrames = loopPeriod;
2591 }
Andy Hung2d85f092015-01-07 12:45:13 -08002592 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002593 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002594 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002595
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002596 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2597 static const uint32_t kPoll = 0;
2598 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2599 minFrames = kPoll * notificationFrames;
2600 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002601
Andy Hunga7f03352015-05-31 21:54:49 -07002602 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2603 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2604 const nsecs_t timeAfterCallbacks = systemTime();
2605
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002606 // Convert frame units to time units
2607 nsecs_t ns = NS_WHENEVER;
2608 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002609 // AudioFlinger consumption of client data may be irregular when coming out of device
2610 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2611 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2612 // half (but no more than half a second) to improve callback accuracy during these temporary
2613 // data surges.
2614 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2615 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2616 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002617 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2618 // TODO: Should we warn if the callback time is too long?
2619 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002620 }
2621
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002622 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2623 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002624 return ns;
2625 }
2626
Andy Hunga7f03352015-05-31 21:54:49 -07002627 // EVENT_MORE_DATA callback handling.
2628 // Timing for linear pcm audio data formats can be derived directly from the
2629 // buffer fill level.
2630 // Timing for compressed data is not directly available from the buffer fill level,
2631 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2632 // to return a certain fill level.
2633
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002634 struct timespec timeout;
2635 const struct timespec *requested = &ClientProxy::kForever;
2636 if (ns != NS_WHENEVER) {
2637 timeout.tv_sec = ns / 1000000000LL;
2638 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002639 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002640 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002641 requested = &timeout;
2642 }
2643
Andy Hungea2b9c02016-02-12 17:06:53 -08002644 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002645 while (mRemainingFrames > 0) {
2646
2647 Buffer audioBuffer;
2648 audioBuffer.frameCount = mRemainingFrames;
2649 size_t nonContig;
2650 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2651 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002652 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002653 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002654 requested = &ClientProxy::kNonBlocking;
2655 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002656 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002657 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002658 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002659 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
2660 (isOffloaded() && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002661 // FIXME bug 25195759
2662 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002663 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002664 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002665 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002666 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002667 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002668
Phil Burkfdb3c072016-02-09 10:47:02 -08002669 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002670 mRetryOnPartialBuffer = false;
2671 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002672 if (ns > 0) { // account for obtain time
2673 const nsecs_t timeNow = systemTime();
2674 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2675 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002676
2677 // delayNs is first computed by the additional frames required in the buffer.
2678 nsecs_t delayNs = framesToNanoseconds(
2679 mRemainingFrames - avail, sampleRate, speed);
2680
2681 // afNs is the AudioFlinger mixer period in ns.
2682 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2683
2684 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2685 // we may have a race if we wait based on the number of frames desired.
2686 // This is a possible issue with resampling and AAudio.
2687 //
2688 // The granularity of audioflinger processing is one mixer period; if
2689 // our wait time is less than one mixer period, wait at most half the period.
2690 if (delayNs < afNs) {
2691 delayNs = std::min(delayNs, afNs / 2);
2692 }
2693
2694 // adjust our ns wait by delayNs.
2695 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2696 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002697 }
2698 return ns;
2699 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002700 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002701
Atneya Nair03079272022-01-18 17:03:14 -05002702 size_t reqSize = audioBuffer.size();
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002703 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2704 // when notifying client it can write more data, pass the total size that can be
2705 // written in the next write() call, since it's not passed through the callback
Atneya Nair03079272022-01-18 17:03:14 -05002706 audioBuffer.mSize += nonContig;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002707 }
Atneya Nairc2dd1272021-10-26 19:39:51 -04002708 const size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002709 ? callback->onMoreData(audioBuffer)
2710 : callback->onCanWriteMoreData(audioBuffer);
Jiabin Huang447cea72020-07-28 22:35:18 +00002711 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002712 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002713 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002714 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002715 return NS_NEVER;
2716 }
2717
2718 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002719 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2720 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2721 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2722 // it only signals to the Java client that it can provide more data, which
2723 // this track is read to accept now.
2724 // The playback thread will be awaken at the next ::write()
2725 return NS_WHENEVER;
2726 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002727 // The callback is done filling buffers
2728 // Keep this thread going to handle timed events and
2729 // still try to get more data in intervals of WAIT_PERIOD_MS
2730 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002731
2732 // mCbf(EVENT_MORE_DATA, ...) might either
2733 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2734 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2735 // (3) Return 0 size when no data is available, does not wait for more data.
2736 //
2737 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2738 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2739 // especially for case (3).
2740 //
2741 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2742 // and this loop; whereas for case (3) we could simply check once with the full
2743 // buffer size and skip the loop entirely.
2744
2745 nsecs_t myns;
Phil Burkfdb3c072016-02-09 10:47:02 -08002746 if (audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002747 // time to wait based on buffer occupancy
2748 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2749 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2750 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002751 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002752 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2753 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2754 myns = datans + (afns / 2);
2755 } else {
2756 // FIXME: This could ping quite a bit if the buffer isn't full.
2757 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2758 myns = kWaitPeriodNs;
2759 }
2760 if (ns > 0) { // account for obtain and callback time
2761 const nsecs_t timeNow = systemTime();
2762 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2763 }
2764 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2765 ns = myns;
2766 }
2767 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002768 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002769
Atneya Nairc2dd1272021-10-26 19:39:51 -04002770 // releaseBuffer reads from audioBuffer.size
Atneya Nair03079272022-01-18 17:03:14 -05002771 audioBuffer.mSize = writtenSize;
Atneya Nairc2dd1272021-10-26 19:39:51 -04002772
Glenn Kasten138d6f92015-03-20 10:54:51 -07002773 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002774 audioBuffer.frameCount = releasedFrames;
2775 mRemainingFrames -= releasedFrames;
2776 if (misalignment >= releasedFrames) {
2777 misalignment -= releasedFrames;
2778 } else {
2779 misalignment = 0;
2780 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002781
2782 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002783 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002784
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002785 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2786 // if callback doesn't like to accept the full chunk
2787 if (writtenSize < reqSize) {
2788 continue;
2789 }
2790
2791 // There could be enough non-contiguous frames available to satisfy the remaining request
2792 if (mRemainingFrames <= nonContig) {
2793 continue;
2794 }
2795
2796#if 0
2797 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2798 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2799 // that total to a sum == notificationFrames.
2800 if (0 < misalignment && misalignment <= mRemainingFrames) {
2801 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002802 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002803 }
2804#endif
2805
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002806 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002807 if (writtenFrames > 0) {
2808 AutoMutex lock(mLock);
2809 mFramesWritten += writtenFrames;
2810 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002811 mRemainingFrames = notificationFrames;
2812 mRetryOnPartialBuffer = true;
2813
2814 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2815 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002816}
2817
Kuowei Li72c8b062023-08-31 13:38:32 +08002818status_t AudioTrack::restoreTrack_l(const char *from, bool forceRestore)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002819{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002820 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2821 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002822 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002823 mediametrics::LogItem(mMetricsId)
2824 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002825 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002826 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2827 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2828 .set(AMEDIAMETRICS_PROP_WHERE, from)
2829 .record(); });
2830
Andy Hungfb8ede22018-09-12 19:03:24 -07002831 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002832 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002833 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002834
Glenn Kastena47f3162012-11-07 10:13:08 -08002835 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002836 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002837 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002838
Kuowei Li72c8b062023-08-31 13:38:32 +08002839 if (!forceRestore &&
2840 (isOffloadedOrDirect_l() || mDoNotReconnect)) {
Andy Hung1f1db832015-06-08 13:26:10 -07002841 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
Atneya Nairb16666a2023-12-11 20:18:33 -08002842 // Disabled since (1) timestamp correction is not implemented for non-PCM and
2843 // (2) We pre-empt existing direct tracks on resource constraint, so these tracks
2844 // shouldn't reconnect.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002845 result = DEAD_OBJECT;
2846 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002847 }
2848
Phil Burk2812d9e2016-01-04 10:34:30 -08002849 // Save so we can return count since creation.
2850 mUnderrunCountOffset = getUnderrunCount_l();
2851
Glenn Kasten200092b2014-08-15 15:13:30 -07002852 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002853 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002854 size_t bufferPosition = 0;
2855 int loopCount = 0;
2856 if (mStaticProxy != 0) {
2857 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002858 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002859 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002860
Andy Hung3c7f47a2021-03-16 17:30:09 -07002861 // save the old startThreshold and framecount
2862 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2863 const uint32_t originalFrameCount = mProxy->frameCount();
2864
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002865 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2866 // causes a lot of churn on the service side, and it can reject starting
2867 // playback of a previously created track. May also apply to other cases.
2868 const int INITIAL_RETRIES = 3;
2869 int retries = INITIAL_RETRIES;
2870retry:
2871 if (retries < INITIAL_RETRIES) {
2872 // See the comment for clearAudioConfigCache at the start of the function.
2873 AudioSystem::clearAudioConfigCache();
2874 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002875 mFlags = mOrigFlags;
2876
Glenn Kasten200092b2014-08-15 15:13:30 -07002877 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002878 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002879 // It will also delete the strong references on previous IAudioTrack and IMemory.
2880 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002881 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002882
Eric Laurent6ec546d2018-10-10 16:52:14 -07002883 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002884 // take the frames that will be lost by track recreation into account in saved position
2885 // For streaming tracks, this is the amount we obtained from the user/client
2886 // (not the number actually consumed at the server - those are already lost).
2887 if (mStaticProxy == 0) {
2888 mPosition = mReleased;
2889 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002890 // Continue playback from last known position and restore loop.
2891 if (mStaticProxy != 0) {
2892 if (loopCount != 0) {
2893 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2894 mLoopStart, mLoopEnd, loopCount);
2895 } else {
2896 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002897 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002898 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002899 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002900 }
2901 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002902 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002903 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2904 sp<VolumeShaper::Operation> operationToEnd =
2905 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002906 // TODO: Ideally we would restore to the exact xOffset position
2907 // as returned by getVolumeShaperState(), but we don't have that
2908 // information when restoring at the client unless we periodically poll
2909 // the server or create shared memory state.
2910 //
Andy Hung39399b62017-04-21 15:07:45 -07002911 // For now, we simply advance to the end of the VolumeShaper effect
2912 // if it has been started.
2913 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002914 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002915 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002916 media::VolumeShaperConfiguration config;
2917 shaper.mConfiguration->writeToParcelable(&config);
2918 media::VolumeShaperOperation operation;
2919 operationToEnd->writeToParcelable(&operation);
2920 status_t status;
2921 mAudioTrack->applyVolumeShaper(config, operation, &status);
2922 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002923 });
2924
Andy Hung3c7f47a2021-03-16 17:30:09 -07002925 // restore the original start threshold if different than frameCount.
2926 if (originalStartThresholdInFrames != originalFrameCount) {
2927 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2928 // and does not trigger a restart.
2929 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
2930 // Any start would be triggered on the mState == ACTIVE check below.
2931 const uint32_t currentThreshold =
2932 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
2933 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
2934 "%s(%d) startThresholdInFrames changing from %u to %u",
2935 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
2936 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002937 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002938 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002939 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002940 // server resets to zero so we offset
2941 mFramesWrittenServerOffset =
2942 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2943 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002944 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002945 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002946 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002947 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002948 // leave time for an eventual race condition to clear before retrying
2949 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002950 goto retry;
2951 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002952 // if no retries left, set invalid bit to force restoring at next occasion
2953 // and avoid inconsistent active state on client and server sides
2954 if (mCblk != nullptr) {
2955 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2956 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002957 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002958 return result;
2959}
2960
Andy Hung90e8a972015-11-09 16:42:40 -08002961Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002962{
2963 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002964 Modulo<uint32_t> newServer(mProxy->getPosition());
2965 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07002966 // TODO There is controversy about whether there can be "negative jitter" in server position.
2967 // This should be investigated further, and if possible, it should be addressed.
2968 // A more definite failure mode is infrequent polling by client.
2969 // One could call (void)getPosition_l() in releaseBuffer(),
2970 // so mReleased and mPosition are always lock-step as best possible.
2971 // That should ensure delta never goes negative for infrequent polling
2972 // unless the server has more than 2^31 frames in its buffer,
2973 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08002974 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07002975 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08002976 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07002977 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08002978 if (delta > 0) { // avoid retrograde
2979 mPosition += delta;
2980 }
2981 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07002982}
2983
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002984bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07002985{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002986 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07002987 // applicable for mixing tracks only (not offloaded or direct)
2988 if (mStaticProxy != 0) {
2989 return true; // static tracks do not have issues with buffer sizing.
2990 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07002991 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08002992 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
2993 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002994 const bool allowed = mFrameCount >= minFrameCount;
2995 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07002996 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002997 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
2998 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002999 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003000 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07003001 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003002 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003003}
3004
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003005status_t AudioTrack::setParameters(const String8& keyValuePairs)
3006{
3007 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003008 status_t status;
3009 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
3010 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003011}
3012
Dean Wheatleya70eef72018-01-04 14:23:50 +11003013status_t AudioTrack::selectPresentation(int presentationId, int programId)
3014{
3015 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08003016 AudioParameter param = AudioParameter();
3017 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
3018 param.addInt(String8(AudioParameter::keyProgramId), programId);
3019 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
Tomasz Wasilczyk8f36f6e2023-08-15 20:59:35 +00003020 __func__, mPortId, param.toString().c_str());
Eric Laurent973db022018-11-20 14:54:31 -08003021
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003022 status_t status;
3023 mAudioTrack->setParameters(param.toString().c_str(), &status);
3024 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11003025}
3026
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003027VolumeShaper::Status AudioTrack::applyVolumeShaper(
3028 const sp<VolumeShaper::Configuration>& configuration,
3029 const sp<VolumeShaper::Operation>& operation)
3030{
3031 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003032 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003033 media::VolumeShaperConfiguration config;
3034 configuration->writeToParcelable(&config);
3035 media::VolumeShaperOperation op;
3036 operation->writeToParcelable(&op);
3037 VolumeShaper::Status status;
3038 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003039
3040 if (status == DEAD_OBJECT) {
3041 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003042 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003043 }
3044 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003045 if (status >= 0) {
3046 // save VolumeShaper for restore
3047 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003048 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3049 mVolumeHandler->setStarted();
3050 }
3051 } else {
3052 // warn only if not an expected restore failure.
3053 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003054 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003055 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003056 return status;
3057}
3058
3059sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3060{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003061 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003062 std::optional<media::VolumeShaperState> vss;
3063 mAudioTrack->getVolumeShaperState(id, &vss);
3064 sp<VolumeShaper::State> state;
3065 if (vss.has_value()) {
3066 state = new VolumeShaper::State();
3067 state->readFromParcelable(vss.value());
3068 }
Andy Hung39399b62017-04-21 15:07:45 -07003069 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3070 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003071 mAudioTrack->getVolumeShaperState(id, &vss);
3072 if (vss.has_value()) {
3073 state = new VolumeShaper::State();
3074 state->readFromParcelable(vss.value());
3075 }
Andy Hung39399b62017-04-21 15:07:45 -07003076 }
3077 }
3078 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003079}
3080
Andy Hungea2b9c02016-02-12 17:06:53 -08003081status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3082{
3083 if (timestamp == nullptr) {
3084 return BAD_VALUE;
3085 }
3086 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003087 return getTimestamp_l(timestamp);
3088}
3089
3090status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3091{
Andy Hungea2b9c02016-02-12 17:06:53 -08003092 if (mCblk->mFlags & CBLK_INVALID) {
3093 const status_t status = restoreTrack_l("getTimestampExtended");
3094 if (status != OK) {
3095 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3096 // recommending that the track be recreated.
3097 return DEAD_OBJECT;
3098 }
3099 }
3100 // check for offloaded/direct here in case restoring somehow changed those flags.
3101 if (isOffloadedOrDirect_l()) {
3102 return INVALID_OPERATION; // not supported
3103 }
3104 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003105 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003106 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003107 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003108 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3109 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3110 // server side frame offset in case AudioTrack has been restored.
3111 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3112 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3113 if (timestamp->mTimeNs[i] >= 0) {
3114 // apply server offset (frames flushed is ignored
3115 // so we don't report the jump when the flush occurs).
3116 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3117 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003118 }
3119 }
3120 return found ? OK : WOULD_BLOCK;
3121}
3122
Glenn Kastence703742013-07-19 16:33:58 -07003123status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3124{
Glenn Kasten53cec222013-08-29 09:01:02 -07003125 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003126 return getTimestamp_l(timestamp);
3127}
Phil Burk1b420972015-04-22 10:52:21 -07003128
Andy Hung65ffdfc2016-10-10 15:52:11 -07003129status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3130{
Phil Burk1b420972015-04-22 10:52:21 -07003131 bool previousTimestampValid = mPreviousTimestampValid;
3132 // Set false here to cover all the error return cases.
3133 mPreviousTimestampValid = false;
3134
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003135 switch (mState) {
3136 case STATE_ACTIVE:
3137 case STATE_PAUSED:
3138 break; // handle below
3139 case STATE_FLUSHED:
3140 case STATE_STOPPED:
3141 return WOULD_BLOCK;
3142 case STATE_STOPPING:
3143 case STATE_PAUSED_STOPPING:
3144 if (!isOffloaded_l()) {
3145 return INVALID_OPERATION;
3146 }
3147 break; // offloaded tracks handled below
3148 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003149 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003150 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003151 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003152 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003153
Eric Laurent275e8e92014-11-30 15:14:47 -08003154 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003155 const status_t status = restoreTrack_l("getTimestamp");
3156 if (status != OK) {
3157 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3158 // recommending that the track be recreated.
3159 return DEAD_OBJECT;
3160 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003161 }
3162
Glenn Kasten200092b2014-08-15 15:13:30 -07003163 // The presented frame count must always lag behind the consumed frame count.
3164 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003165
3166 status_t status;
jiabin94ed47c2023-07-27 23:34:20 +00003167 if (isAfTrackOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003168 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003169 media::AudioTimestampInternal ts;
3170 mAudioTrack->getTimestamp(&ts, &status);
3171 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003172 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003173 }
Andy Hung6ae58432016-02-16 18:32:24 -08003174 } else {
3175 // read timestamp from shared memory
3176 ExtendedTimestamp ets;
3177 status = mProxy->getTimestamp(&ets);
3178 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003179 ExtendedTimestamp::Location location;
3180 status = ets.getBestTimestamp(&timestamp, &location);
3181
3182 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003183 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003184 // It is possible that the best location has moved from the kernel to the server.
3185 // In this case we adjust the position from the previous computed latency.
3186 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3187 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003188 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003189 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003190 // check that the last kernel OK time info exists and the positions
3191 // are valid (if they predate the current track, the positions may
3192 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003193 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003194 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003195 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3196 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3197 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003198 ?
3199 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3200 / 1000)
3201 :
3202 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3203 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003204 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003205 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003206 if (frames >= ets.mPosition[location]) {
3207 timestamp.mPosition = 0;
3208 } else {
3209 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3210 }
Andy Hung69488c42016-05-16 18:43:33 -07003211 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3212 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003213 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003214 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003215
3216 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3217 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3218 // In Q, we don't return errors as an invalid time
3219 // but instead we leave the last kernel good timestamp alone.
3220 //
3221 // If server is identical to kernel, the device data pipeline is idle.
3222 // A better start time is now. The retrograde check ensures
3223 // timestamp monotonicity.
3224 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003225 if (!mTimestampStallReported) {
3226 ALOGD("%s(%d): device stall time corrected using current time %lld",
3227 __func__, mPortId, (long long)nowNs);
3228 mTimestampStallReported = true;
3229 }
Andy Hung98731a22019-04-08 19:19:07 -07003230 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003231 } else {
3232 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003233 }
Andy Hungb01faa32016-04-27 12:51:32 -07003234 }
Andy Hung5d313802016-10-10 15:09:39 -07003235
3236 // We update the timestamp time even when paused.
3237 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3238 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003239 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003240 const int64_t lag =
3241 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3242 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3243 ? int64_t(mAfLatency * 1000000LL)
3244 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3245 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3246 * NANOS_PER_SECOND / mSampleRate;
3247 const int64_t limit = now - lag; // no earlier than this limit
3248 if (at < limit) {
3249 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3250 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003251 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003252 }
3253 }
Andy Hungb01faa32016-04-27 12:51:32 -07003254 mPreviousLocation = location;
3255 } else {
3256 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003257 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003258 }
Andy Hung6ae58432016-02-16 18:32:24 -08003259 }
3260 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003261 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3262 // other failures are signaled by a negative time.
3263 // If we come out of FLUSHED or STOPPED where the position is known
3264 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3265 // "zero" for NuPlayer). We don't convert for track restoration as position
3266 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003267 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003268 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003269 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3270 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3271 status = WOULD_BLOCK;
3272 }
Andy Hung6ae58432016-02-16 18:32:24 -08003273 }
3274 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003275 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003276 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003277 return status;
3278 }
jiabin94ed47c2023-07-27 23:34:20 +00003279 if (isAfTrackOffloadedOrDirect_l()) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003280 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3281 // use cached paused position in case another offloaded track is running.
3282 timestamp.mPosition = mPausedPosition;
3283 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003284 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003285 return NO_ERROR;
3286 }
3287
3288 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003289 // be asynchronous or return near finish or exhibit glitchy behavior.
3290 //
3291 // Originally this showed up as the first timestamp being a continuation of
3292 // the previous song under gapless playback.
3293 // However, we sometimes see zero timestamps, then a glitch of
3294 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003295 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003296 static const int kTimeJitterUs = 100000; // 100 ms
3297 static const int k1SecUs = 1000000;
3298
3299 const int64_t timeNow = getNowUs();
3300
Andy Hungffa36952017-08-17 10:41:51 -07003301 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003302 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003303 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003304 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3305 }
Andy Hungffa36952017-08-17 10:41:51 -07003306 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003307 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003308 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003309
3310 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3311 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003312 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003313 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003314 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003315 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003316 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003317 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003318 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3319 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003320 mTimestampStartupGlitchReported = true;
3321 if (previousTimestampValid
3322 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3323 timestamp = mPreviousTimestamp;
3324 mPreviousTimestampValid = true;
3325 return NO_ERROR;
3326 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003327 return WOULD_BLOCK;
3328 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003329 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003330 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003331 }
3332 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003333 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003334 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003335 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003336 }
3337 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003338 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3339 (void) updateAndGetPosition_l();
3340 // Server consumed (mServer) and presented both use the same server time base,
3341 // and server consumed is always >= presented.
3342 // The delta between these represents the number of frames in the buffer pipeline.
3343 // If this delta between these is greater than the client position, it means that
3344 // actually presented is still stuck at the starting line (figuratively speaking),
3345 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003346 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3347 // mPosition exceeds 32 bits.
3348 // TODO Remove when timestamp is updated to contain pipeline status info.
3349 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3350 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3351 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003352 return INVALID_OPERATION;
3353 }
3354 // Convert timestamp position from server time base to client time base.
3355 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3356 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003357 // Use Modulo computation here.
3358 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003359 // Immediately after a call to getPosition_l(), mPosition and
3360 // mServer both represent the same frame position. mPosition is
3361 // in client's point of view, and mServer is in server's point of
3362 // view. So the difference between them is the "fudge factor"
3363 // between client and server views due to stop() and/or new
3364 // IAudioTrack. And timestamp.mPosition is initially in server's
3365 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003366 }
Phil Burk1b420972015-04-22 10:52:21 -07003367
3368 // Prevent retrograde motion in timestamp.
3369 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3370 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003371 // Fix stale time when checking timestamp right after start().
3372 // The position is at the last reported location but the time can be stale
3373 // due to pause or standby or cold start latency.
3374 //
3375 // We keep advancing the time (but not the position) to ensure that the
3376 // stale value does not confuse the application.
3377 //
3378 // For offload compatibility, use a default lag value here.
3379 // Any time discrepancy between this update and the pause timestamp is handled
3380 // by the retrograde check afterwards.
3381 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3382 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3383 const int64_t limitNs = mStartNs - lagNs;
3384 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003385 if (!mTimestampStaleTimeReported) {
3386 ALOGD("%s(%d): stale timestamp time corrected, "
3387 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3388 __func__, mPortId,
3389 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3390 mTimestampStaleTimeReported = true;
3391 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003392 timestamp.mTime = convertNsToTimespec(limitNs);
3393 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003394 } else {
3395 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003396 }
3397
Andy Hungffa36952017-08-17 10:41:51 -07003398 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003399 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003400 const int64_t previousTimeNanos =
3401 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003402
3403 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003404 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003405 if (!mTimestampRetrogradeTimeReported) {
3406 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3407 __func__, mPortId,
3408 (long long)currentTimeNanos, (long long)previousTimeNanos);
3409 mTimestampRetrogradeTimeReported = true;
3410 }
Andy Hung5d313802016-10-10 15:09:39 -07003411 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003412 } else {
3413 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003414 }
3415
3416 // Looking at signed delta will work even when the timestamps
3417 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003418 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3419 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003420 if (deltaPosition < 0) {
3421 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003422 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003423 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003424 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003425 deltaPosition,
3426 timestamp.mPosition,
3427 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003428 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003429 }
3430 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003431 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003432 }
Andy Hung5d313802016-10-10 15:09:39 -07003433 if (deltaPosition < 0) {
3434 timestamp.mPosition = mPreviousTimestamp.mPosition;
3435 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003436 }
Andy Hung5d313802016-10-10 15:09:39 -07003437#if 0
3438 // Uncomment this to verify audio timestamp rate.
3439 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003440 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003441 if (deltaTime != 0) {
3442 const int64_t computedSampleRate =
3443 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003444 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003445 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003446 (unsigned)computedSampleRate, mSampleRate);
3447 }
3448#endif
Phil Burk1b420972015-04-22 10:52:21 -07003449 }
3450 mPreviousTimestamp = timestamp;
3451 mPreviousTimestampValid = true;
3452 }
3453
Glenn Kastenfe346c72013-08-30 13:28:22 -07003454 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003455}
3456
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003457String8 AudioTrack::getParameters(const String8& keys)
3458{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003459 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003460 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003461 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003462 } else {
Tomasz Wasilczyk53ce3af2023-08-14 16:16:55 +00003463 return String8();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003464 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003465}
3466
Glenn Kasten23a75452014-01-13 10:37:17 -08003467bool AudioTrack::isOffloaded() const
3468{
3469 AutoMutex lock(mLock);
3470 return isOffloaded_l();
3471}
3472
Eric Laurentab5cdba2014-06-09 17:22:27 -07003473bool AudioTrack::isDirect() const
3474{
3475 AutoMutex lock(mLock);
3476 return isDirect_l();
3477}
3478
3479bool AudioTrack::isOffloadedOrDirect() const
3480{
3481 AutoMutex lock(mLock);
3482 return isOffloadedOrDirect_l();
3483}
3484
3485
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003486status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003487{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003488 String8 result;
3489
3490 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003491 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003492 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003493 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003494 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003495 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003496 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003497 mFormat, mChannelMask, mChannelCount);
3498 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3499 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3500 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3501 mFrameCount, mReqFrameCount);
3502 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3503 " req. notif. per buff(%u)\n",
3504 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3505 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3506 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3507 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3508 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
Tomasz Wasilczyk8f36f6e2023-08-15 20:59:35 +00003509 ::write(fd, result.c_str(), result.size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003510 return NO_ERROR;
3511}
3512
Phil Burk2812d9e2016-01-04 10:34:30 -08003513uint32_t AudioTrack::getUnderrunCount() const
3514{
3515 AutoMutex lock(mLock);
3516 return getUnderrunCount_l();
3517}
3518
3519uint32_t AudioTrack::getUnderrunCount_l() const
3520{
3521 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3522}
3523
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003524uint32_t AudioTrack::getUnderrunFrames() const
3525{
3526 AutoMutex lock(mLock);
3527 return mProxy->getUnderrunFrames();
3528}
3529
Andy Hung3a5c2f32021-02-17 15:06:42 -08003530void AudioTrack::setLogSessionId(const char *logSessionId)
3531{
3532 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003533 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003534 if (mLogSessionId == logSessionId) return;
3535
3536 mLogSessionId = logSessionId;
3537 mediametrics::LogItem(mMetricsId)
3538 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3539 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3540 .record();
3541}
3542
Andy Hung839a3062021-02-17 11:15:16 -08003543void AudioTrack::setPlayerIId(int playerIId)
3544{
3545 AutoMutex lock(mLock);
3546 if (mPlayerIId == playerIId) return;
3547
3548 mPlayerIId = playerIId;
Vlad Popaad0fe922022-06-10 00:43:14 +02003549 triggerPortIdUpdate_l();
Andy Hung839a3062021-02-17 11:15:16 -08003550 mediametrics::LogItem(mMetricsId)
3551 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3552 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3553 .record();
3554}
3555
Vlad Popaad0fe922022-06-10 00:43:14 +02003556void AudioTrack::triggerPortIdUpdate_l() {
3557 if (mAudioManager == nullptr) {
3558 // use checkService() to avoid blocking if audio service is not up yet
3559 sp<IBinder> binder =
3560 defaultServiceManager()->checkService(String16(kAudioServiceName));
3561 if (binder == nullptr) {
3562 ALOGE("%s(%d): binding to audio service failed.",
3563 __func__,
3564 mPlayerIId);
3565 return;
3566 }
3567
3568 mAudioManager = interface_cast<IAudioManager>(binder);
3569 }
3570
3571 // first time when the track is created we do not have a valid piid
3572 if (mPlayerIId != PLAYER_PIID_INVALID) {
3573 mAudioManager->playerEvent(mPlayerIId, PLAYER_UPDATE_PORT_ID, mPortId);
3574 }
3575}
3576
Eric Laurent296fb132015-05-01 11:38:42 -07003577status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3578{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003579
Eric Laurent296fb132015-05-01 11:38:42 -07003580 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003581 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003582 return BAD_VALUE;
3583 }
3584 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003585 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003586 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003587 return INVALID_OPERATION;
3588 }
3589 status_t status = NO_ERROR;
3590 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3591 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003592 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003593 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003594 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003595 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003596 }
3597 mDeviceCallback = callback;
3598 return status;
3599}
3600
3601status_t AudioTrack::removeAudioDeviceCallback(
3602 const sp<AudioSystem::AudioDeviceCallback>& callback)
3603{
3604 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003605 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003606 return BAD_VALUE;
3607 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003608 AutoMutex lock(mLock);
3609 if (mDeviceCallback.unsafe_get() != callback.get()) {
3610 ALOGW("%s removing different callback!", __FUNCTION__);
3611 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003612 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003613 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003614 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003615 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003616 }
Eric Laurent296fb132015-05-01 11:38:42 -07003617 return NO_ERROR;
3618}
3619
Eric Laurentad2e7b92017-09-14 20:06:42 -07003620
3621void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3622 audio_port_handle_t deviceId)
3623{
3624 sp<AudioSystem::AudioDeviceCallback> callback;
3625 {
3626 AutoMutex lock(mLock);
3627 if (audioIo != mOutput) {
3628 return;
3629 }
3630 callback = mDeviceCallback.promote();
3631 // only update device if the track is active as route changes due to other use cases are
3632 // irrelevant for this client
3633 if (mState == STATE_ACTIVE) {
3634 mRoutedDeviceId = deviceId;
3635 }
3636 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003637
Eric Laurentad2e7b92017-09-14 20:06:42 -07003638 if (callback.get() != nullptr) {
3639 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3640 }
3641}
3642
Andy Hunge13f8a62016-03-30 14:20:42 -07003643status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3644{
3645 if (msec == nullptr ||
3646 (location != ExtendedTimestamp::LOCATION_SERVER
3647 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3648 return BAD_VALUE;
3649 }
3650 AutoMutex lock(mLock);
3651 // inclusive of offloaded and direct tracks.
3652 //
3653 // It is possible, but not enabled, to allow duration computation for non-pcm
3654 // audio_has_proportional_frames() formats because currently they have
3655 // the drain rate equivalent to the pcm sample rate * framesize.
3656 if (!isPurePcmData_l()) {
3657 return INVALID_OPERATION;
3658 }
3659 ExtendedTimestamp ets;
3660 if (getTimestamp_l(&ets) == OK
3661 && ets.mTimeNs[location] > 0) {
3662 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3663 - ets.mPosition[location];
3664 if (diff < 0) {
3665 *msec = 0;
3666 } else {
3667 // ms is the playback time by frames
3668 int64_t ms = (int64_t)((double)diff * 1000 /
3669 ((double)mSampleRate * mPlaybackRate.mSpeed));
3670 // clockdiff is the timestamp age (negative)
3671 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3672 ets.mTimeNs[location]
3673 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3674 - systemTime(SYSTEM_TIME_MONOTONIC);
3675
3676 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3677 static const int NANOS_PER_MILLIS = 1000000;
3678 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3679 }
3680 return NO_ERROR;
3681 }
3682 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3683 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3684 }
3685 // use server position directly (offloaded and direct arrive here)
3686 updateAndGetPosition_l();
3687 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3688 *msec = (diff <= 0) ? 0
3689 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3690 return NO_ERROR;
3691}
3692
Andy Hung65ffdfc2016-10-10 15:52:11 -07003693bool AudioTrack::hasStarted()
3694{
3695 AutoMutex lock(mLock);
3696 switch (mState) {
3697 case STATE_STOPPED:
3698 if (isOffloadedOrDirect_l()) {
3699 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003700 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003701 }
3702 // A normal audio track may still be draining, so
3703 // check if stream has ended. This covers fasttrack position
3704 // instability and start/stop without any data written.
3705 if (mProxy->getStreamEndDone()) {
3706 return true;
3707 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003708 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003709 case STATE_ACTIVE:
3710 case STATE_STOPPING:
3711 break;
3712 case STATE_PAUSED:
3713 case STATE_PAUSED_STOPPING:
3714 case STATE_FLUSHED:
3715 return false; // we're not active
3716 default:
Eric Laurent973db022018-11-20 14:54:31 -08003717 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003718 break;
3719 }
3720
3721 // wait indicates whether we need to wait for a timestamp.
3722 // This is conservatively figured - if we encounter an unexpected error
3723 // then we will not wait.
3724 bool wait = false;
jiabin94ed47c2023-07-27 23:34:20 +00003725 if (isAfTrackOffloadedOrDirect_l()) {
Andy Hung65ffdfc2016-10-10 15:52:11 -07003726 AudioTimestamp ts;
3727 status_t status = getTimestamp_l(ts);
3728 if (status == WOULD_BLOCK) {
3729 wait = true;
3730 } else if (status == OK) {
3731 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3732 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003733 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003734 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003735 (int)wait,
3736 ts.mPosition,
3737 (long long)mStartTs.mPosition);
3738 } else {
3739 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3740 ExtendedTimestamp ets;
3741 status_t status = getTimestamp_l(&ets);
3742 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3743 wait = true;
3744 } else if (status == OK) {
3745 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3746 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3747 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3748 continue;
3749 }
3750 wait = ets.mPosition[location] == 0
3751 || ets.mPosition[location] == mStartEts.mPosition[location];
3752 break;
3753 }
3754 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003755 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003756 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003757 (int)wait,
3758 (long long)ets.mPosition[location],
3759 (long long)mStartEts.mPosition[location]);
3760 }
3761 return !wait;
3762}
3763
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003764// =========================================================================
3765
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003766void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003767{
3768 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3769 if (audioTrack != 0) {
3770 AutoMutex lock(audioTrack->mLock);
3771 audioTrack->mProxy->binderDied();
3772 }
3773}
3774
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003775// =========================================================================
3776
Andy Hungca353672019-03-06 11:54:38 -08003777AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003778 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3779 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003780 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003781{
3782}
3783
3784AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003785{
3786}
3787
3788bool AudioTrack::AudioTrackThread::threadLoop()
3789{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003790 {
3791 AutoMutex _l(mMyLock);
3792 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003793 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003794 mMyCond.wait(mMyLock);
3795 // caller will check for exitPending()
3796 return true;
3797 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003798 if (mIgnoreNextPausedInt) {
3799 mIgnoreNextPausedInt = false;
3800 mPausedInt = false;
3801 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003802 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003803 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003804 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003805 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003806 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3807 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003808 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003809 mMyCond.wait(mMyLock);
3810 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003811 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003812 return true;
3813 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003814 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003815 if (exitPending()) {
3816 return false;
3817 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003818 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003819 switch (ns) {
3820 case 0:
3821 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003822 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003823 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003824 return true;
3825 case NS_NEVER:
3826 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003827 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003828 // Event driven: call wake() when callback notifications conditions change.
3829 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003830 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003831 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003832 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003833 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003834 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003835 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003836 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003837}
3838
Glenn Kasten3acbd052012-02-28 10:39:56 -08003839void AudioTrack::AudioTrackThread::requestExit()
3840{
3841 // must be in this order to avoid a race condition
3842 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003843 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003844}
3845
3846void AudioTrack::AudioTrackThread::pause()
3847{
3848 AutoMutex _l(mMyLock);
3849 mPaused = true;
3850}
3851
3852void AudioTrack::AudioTrackThread::resume()
3853{
3854 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003855 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003856 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003857 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003858 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003859 mMyCond.signal();
3860 }
3861}
3862
Andy Hung3c09c782014-12-29 18:39:32 -08003863void AudioTrack::AudioTrackThread::wake()
3864{
3865 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003866 if (!mPaused) {
3867 // wake() might be called while servicing a callback - ignore the next
3868 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003869 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003870 if (mPausedInt && mPausedNs > 0) {
3871 // audio track is active and internally paused with timeout.
3872 mPausedInt = false;
3873 mMyCond.signal();
3874 }
Andy Hung3c09c782014-12-29 18:39:32 -08003875 }
3876}
3877
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003878void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3879{
3880 AutoMutex _l(mMyLock);
3881 mPausedInt = true;
3882 mPausedNs = ns;
3883}
3884
jiabinf6eb4c32020-02-25 14:06:25 -08003885binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3886 const std::vector<uint8_t>& audioMetadata)
3887{
3888 AutoMutex _l(mAudioTrackCbLock);
3889 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3890 if (callback.get() != nullptr) {
3891 callback->onCodecFormatChanged(audioMetadata);
3892 } else {
3893 mCallback.clear();
3894 }
3895 return binder::Status::ok();
3896}
3897
3898void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3899 const sp<media::IAudioTrackCallback> &callback) {
3900 AutoMutex lock(mAudioTrackCbLock);
3901 mCallback = callback;
3902}
3903
Glenn Kasten40bc9062015-03-20 09:09:33 -07003904} // namespace android