blob: 807aa13e56ea1452a49a12cb8db87539e74ec8c0 [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>
Mark Salyzyn34fb2962014-06-18 16:30:56 -070024
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -070025#include <android-base/macros.h>
Andy Hung2b01f002017-07-05 12:01:36 -070026#include <audio_utils/clock.h>
Glenn Kasten9f80dd22012-12-18 15:57:32 -080027#include <audio_utils/primitives.h>
28#include <binder/IPCThreadState.h>
29#include <media/AudioTrack.h>
30#include <utils/Log.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080031#include <private/media/AudioTrackShared.h>
Suren Baghdasaryan7435e7d2018-12-19 17:09:28 -080032#include <processgroup/sched_policy.h>
Glenn Kasten1ab85ec2013-05-31 09:18:43 -070033#include <media/IAudioFlinger.h>
Michael Chana94fbb22018-04-24 14:31:19 +100034#include <media/IAudioPolicyService.h>
Dean Wheatleya70eef72018-01-04 14:23:50 +110035#include <media/AudioParameter.h>
Andy Hungcd044842014-08-07 11:04:34 -070036#include <media/AudioResamplerPublic.h>
Michael Chana94fbb22018-04-24 14:31:19 +100037#include <media/AudioSystem.h>
Ray Essickf27e9872019-12-07 06:28:46 -080038#include <media/MediaMetricsItem.h>
Ray Essicked304702017-12-12 14:00:57 -080039#include <media/TypeConverter.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080040
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +010041#define WAIT_PERIOD_MS 10
42#define WAIT_STREAM_END_TIMEOUT_SEC 120
Andy Hung53c3b5f2014-12-15 16:42:05 -080043static const int kMaxLoopCountNotifications = 32;
Glenn Kasten511754b2012-01-11 09:52:19 -080044
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080045namespace android {
Chia-chi Yeh33005a92010-06-16 06:33:13 +080046// ---------------------------------------------------------------------------
47
Ivan Lozano8cf3a072017-08-09 09:01:33 -070048using media::VolumeShaper;
49
Andy Hunga7f03352015-05-31 21:54:49 -070050// TODO: Move to a separate .h
51
Andy Hung4ede21d2014-12-12 15:37:34 -080052template <typename T>
Andy Hunga7f03352015-05-31 21:54:49 -070053static inline const T &min(const T &x, const T &y) {
Andy Hung4ede21d2014-12-12 15:37:34 -080054 return x < y ? x : y;
55}
56
Andy Hunga7f03352015-05-31 21:54:49 -070057template <typename T>
58static inline const T &max(const T &x, const T &y) {
59 return x > y ? x : y;
60}
61
62static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
63{
64 return ((double)frames * 1000000000) / ((double)sampleRate * speed);
65}
66
Andy Hung7f1bc8a2014-09-12 14:43:11 -070067static int64_t convertTimespecToUs(const struct timespec &tv)
68{
Chih-Hung Hsieh50fa06c2018-12-11 13:51:36 -080069 return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -070070}
71
Andy Hungffa36952017-08-17 10:41:51 -070072// TODO move to audio_utils.
73static inline struct timespec convertNsToTimespec(int64_t ns) {
74 struct timespec tv;
75 tv.tv_sec = static_cast<time_t>(ns / NANOS_PER_SECOND);
Andy Hung06a730b2020-04-09 13:28:31 -070076 tv.tv_nsec = static_cast<int64_t>(ns % NANOS_PER_SECOND);
Andy Hungffa36952017-08-17 10:41:51 -070077 return tv;
78}
79
Andy Hung7f1bc8a2014-09-12 14:43:11 -070080// current monotonic time in microseconds.
81static int64_t getNowUs()
82{
83 struct timespec tv;
84 (void) clock_gettime(CLOCK_MONOTONIC, &tv);
85 return convertTimespecToUs(tv);
86}
87
Andy Hung26145642015-04-15 21:56:53 -070088// FIXME: we don't use the pitch setting in the time stretcher (not working);
89// instead we emulate it using our sample rate converter.
90static const bool kFixPitch = true; // enable pitch fix
91static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)
92{
93 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate;
94}
95
96static inline float adjustSpeed(float speed, float pitch)
97{
Ricardo Garcia6c7f0622015-04-30 18:39:16 -070098 return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
Andy Hung26145642015-04-15 21:56:53 -070099}
100
101static inline float adjustPitch(float pitch)
102{
103 return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
104}
105
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800106// static
107status_t AudioTrack::getMinFrameCount(
Glenn Kastene33054e2012-11-14 12:54:39 -0800108 size_t* frameCount,
Glenn Kastenfff6d712012-01-12 16:38:12 -0800109 audio_stream_type_t streamType,
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800110 uint32_t sampleRate)
111{
Glenn Kastend65d73c2012-06-22 17:21:07 -0700112 if (frameCount == NULL) {
113 return BAD_VALUE;
114 }
Glenn Kasten04cd0182012-06-25 11:49:27 -0700115
Andy Hung0e48d252015-01-26 11:43:15 -0800116 // FIXME handle in server, like createTrack_l(), possible missing info:
Glenn Kastene0fa4672012-04-24 14:35:14 -0700117 // audio_io_handle_t output
118 // audio_format_t format
119 // audio_channel_mask_t channelMask
Andy Hung0e48d252015-01-26 11:43:15 -0800120 // audio_output_flags_t flags (FAST)
Glenn Kasten3b16c762012-11-14 08:44:39 -0800121 uint32_t afSampleRate;
Glenn Kasten66a04672014-01-08 08:53:44 -0800122 status_t status;
123 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
124 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700125 ALOGE("%s(): Unable to query output sample rate for stream type %d; status %d",
126 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800127 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800128 }
Glenn Kastene33054e2012-11-14 12:54:39 -0800129 size_t afFrameCount;
Glenn Kasten66a04672014-01-08 08:53:44 -0800130 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
131 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700132 ALOGE("%s(): Unable to query output frame count for stream type %d; status %d",
133 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800134 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800135 }
136 uint32_t afLatency;
Glenn Kasten66a04672014-01-08 08:53:44 -0800137 status = AudioSystem::getOutputLatency(&afLatency, streamType);
138 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700139 ALOGE("%s(): Unable to query output latency for stream type %d; status %d",
140 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800141 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800142 }
143
Andy Hung8edb8dc2015-03-26 19:13:55 -0700144 // When called from createTrack, speed is 1.0f (normal speed).
145 // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
Eric Laurent21da6472017-11-09 16:29:26 -0800146 *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
147 sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800148
Andy Hung0e48d252015-01-26 11:43:15 -0800149 // The formula above should always produce a non-zero value under normal circumstances:
150 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
151 // 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 -0800152 if (*frameCount == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700153 ALOGE("%s(): failed for streamType %d, sampleRate %u",
154 __func__, streamType, sampleRate);
Glenn Kasten66a04672014-01-08 08:53:44 -0800155 return BAD_VALUE;
156 }
Andy Hungfb8ede22018-09-12 19:03:24 -0700157 ALOGV("%s(): getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
158 __func__, *frameCount, afFrameCount, afSampleRate, afLatency);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800159 return NO_ERROR;
160}
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800161
Michael Chana94fbb22018-04-24 14:31:19 +1000162// static
163bool AudioTrack::isDirectOutputSupported(const audio_config_base_t& config,
164 const audio_attributes_t& attributes) {
165 ALOGV("%s()", __FUNCTION__);
166 const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
167 if (aps == 0) return false;
168 return aps->isDirectOutputSupported(config, attributes);
169}
170
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800171// ---------------------------------------------------------------------------
172
Ray Essicked304702017-12-12 14:00:57 -0800173void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
174{
Ray Essick88394302018-01-24 14:52:05 -0800175 // only if we're in a good state...
176 // XXX: shall we gather alternative info if failing?
177 const status_t lstatus = track->initCheck();
178 if (lstatus != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700179 ALOGD("%s(): no metrics gathered, track status=%d", __func__, (int) lstatus);
Ray Essick88394302018-01-24 14:52:05 -0800180 return;
181 }
182
Andy Hungd0979812019-02-21 15:51:44 -0800183#define MM_PREFIX "android.media.audiotrack." // avoid cut-n-paste errors.
Ray Essicked304702017-12-12 14:00:57 -0800184
Andy Hungd0979812019-02-21 15:51:44 -0800185 // Java API 28 entries, do not change.
Ray Essickf27e9872019-12-07 06:28:46 -0800186 mMetricsItem->setCString(MM_PREFIX "streamtype", toString(track->streamType()).c_str());
187 mMetricsItem->setCString(MM_PREFIX "type",
Andy Hungd0979812019-02-21 15:51:44 -0800188 toString(track->mAttributes.content_type).c_str());
Ray Essickf27e9872019-12-07 06:28:46 -0800189 mMetricsItem->setCString(MM_PREFIX "usage", toString(track->mAttributes.usage).c_str());
Ray Essicked304702017-12-12 14:00:57 -0800190
Andy Hungd0979812019-02-21 15:51:44 -0800191 // Non-API entries, these can change due to a Java string mistake.
Ray Essickf27e9872019-12-07 06:28:46 -0800192 mMetricsItem->setInt32(MM_PREFIX "sampleRate", (int32_t)track->mSampleRate);
193 mMetricsItem->setInt64(MM_PREFIX "channelMask", (int64_t)track->mChannelMask);
Andy Hungd0979812019-02-21 15:51:44 -0800194 // Non-API entries, these can change.
Ray Essickf27e9872019-12-07 06:28:46 -0800195 mMetricsItem->setInt32(MM_PREFIX "portId", (int32_t)track->mPortId);
196 mMetricsItem->setCString(MM_PREFIX "encoding", toString(track->mFormat).c_str());
197 mMetricsItem->setInt32(MM_PREFIX "frameCount", (int32_t)track->mFrameCount);
198 mMetricsItem->setCString(MM_PREFIX "attributes", toString(track->mAttributes).c_str());
Ray Essicked304702017-12-12 14:00:57 -0800199}
200
Ray Essick88394302018-01-24 14:52:05 -0800201// hand the user a snapshot of the metrics.
Ray Essickf27e9872019-12-07 06:28:46 -0800202status_t AudioTrack::getMetrics(mediametrics::Item * &item)
Ray Essick88394302018-01-24 14:52:05 -0800203{
204 mMediaMetrics.gather(this);
Ray Essickf27e9872019-12-07 06:28:46 -0800205 mediametrics::Item *tmp = mMediaMetrics.dup();
Ray Essick88394302018-01-24 14:52:05 -0800206 if (tmp == nullptr) {
207 return BAD_VALUE;
208 }
209 item = tmp;
210 return NO_ERROR;
211}
Ray Essicked304702017-12-12 14:00:57 -0800212
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800213AudioTrack::AudioTrack()
Glenn Kasten87913512011-06-22 16:15:25 -0700214 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700215 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800216 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800217 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700218 mPausedPosition(0),
Eric Laurent20b9ef02016-12-05 11:03:16 -0800219 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
jiabinf6eb4c32020-02-25 14:06:25 -0800220 mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
221 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800222{
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700223 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
224 mAttributes.usage = AUDIO_USAGE_UNKNOWN;
225 mAttributes.flags = 0x0;
226 strcpy(mAttributes.tags, "");
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800227}
228
229AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800230 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800231 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800232 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700233 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800234 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700235 audio_output_flags_t flags,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800236 callback_t cbf,
237 void* user,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700238 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800239 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000240 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800241 const audio_offload_info_t *offloadInfo,
Andy Hung1f12a8a2016-11-07 16:10:30 -0800242 uid_t uid,
Jean-Michel Trivid9d7fa02014-06-24 08:01:46 -0700243 pid_t pid,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700244 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700245 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700246 float maxRequiredSpeed,
247 audio_port_handle_t selectedDeviceId)
Glenn Kasten87913512011-06-22 16:15:25 -0700248 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700249 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800250 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800251 mPreviousSchedulingGroup(SP_DEFAULT),
jiabinf6eb4c32020-02-25 14:06:25 -0800252 mPausedPosition(0),
253 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800254{
François Gaffie393f0e02019-04-10 09:09:08 +0200255 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900256
Eric Laurentf32d7812017-11-30 14:44:07 -0800257 (void)set(streamType, sampleRate, format, channelMask,
Eric Laurenta514bdb2010-06-21 09:27:30 -0700258 frameCount, flags, cbf, user, notificationFrames,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800259 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
jiabin156c6872017-10-06 09:47:15 -0700260 offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800261}
262
Andreas Huberc8139852012-01-18 10:51:55 -0800263AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800264 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800265 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800266 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700267 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800268 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700269 audio_output_flags_t flags,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800270 callback_t cbf,
271 void* user,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700272 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800273 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000274 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800275 const audio_offload_info_t *offloadInfo,
Andy Hung1f12a8a2016-11-07 16:10:30 -0800276 uid_t uid,
Jean-Michel Trivid9d7fa02014-06-24 08:01:46 -0700277 pid_t pid,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700278 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700279 bool doNotReconnect,
280 float maxRequiredSpeed)
Glenn Kasten87913512011-06-22 16:15:25 -0700281 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700282 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800283 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800284 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700285 mPausedPosition(0),
jiabinf6eb4c32020-02-25 14:06:25 -0800286 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
287 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800288{
François Gaffie393f0e02019-04-10 09:09:08 +0200289 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900290
Eric Laurentf32d7812017-11-30 14:44:07 -0800291 (void)set(streamType, sampleRate, format, channelMask,
Glenn Kasten17a736c2012-02-14 08:52:15 -0800292 0 /*frameCount*/, flags, cbf, user, notificationFrames,
Marco Nelissend457c972014-02-11 08:47:07 -0800293 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
Andy Hungff874dc2016-04-11 16:49:09 -0700294 uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800295}
296
297AudioTrack::~AudioTrack()
298{
Ray Essicked304702017-12-12 14:00:57 -0800299 // pull together the numbers, before we clean up our structures
300 mMediaMetrics.gather(this);
301
Andy Hungb68f5eb2019-12-03 16:49:17 -0800302 mediametrics::LogItem(mMetricsId)
303 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700304 .set(AMEDIAMETRICS_PROP_CALLERNAME,
305 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700306 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700307 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800308 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
309 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
310 .record();
311
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800312 if (mStatus == NO_ERROR) {
313 // Make sure that callback function exits in the case where
314 // it is looping on buffer full condition in obtainBuffer().
315 // Otherwise the callback thread will never exit.
316 stop();
317 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100318 mProxy->interrupt();
Glenn Kasten3acbd052012-02-28 10:39:56 -0800319 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800320 mAudioTrackThread->requestExitAndWait();
321 mAudioTrackThread.clear();
322 }
Eric Laurent296fb132015-05-01 11:38:42 -0700323 // No lock here: worst case we remove a NULL callback which will be a nop
324 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -0700325 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -0700326 }
Marco Nelissenf8880202014-11-14 07:58:25 -0800327 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700328 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700329 mCblkMemory.clear();
330 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800331 IPCThreadState::self()->flushCommands();
Andy Hungfb8ede22018-09-12 19:03:24 -0700332 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800333 __func__, mPortId,
Glenn Kasten4c36d6f2015-03-20 09:05:01 -0700334 mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid);
Marco Nelissend457c972014-02-11 08:47:07 -0800335 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800336 }
337}
338
339status_t AudioTrack::set(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800340 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800341 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800342 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700343 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800344 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700345 audio_output_flags_t flags,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800346 callback_t cbf,
347 void* user,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700348 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800349 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700350 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800351 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000352 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800353 const audio_offload_info_t *offloadInfo,
Andy Hung1f12a8a2016-11-07 16:10:30 -0800354 uid_t uid,
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700355 pid_t pid,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700356 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700357 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700358 float maxRequiredSpeed,
359 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800360{
Eric Laurentf32d7812017-11-30 14:44:07 -0800361 status_t status;
362 uint32_t channelCount;
363 pid_t callingPid;
364 pid_t myPid;
365
Eric Laurent973db022018-11-20 14:54:31 -0800366 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700367 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
Glenn Kastenea38ee72016-04-18 11:08:01 -0700368 "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700369 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800370 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Glenn Kasten4c36d6f2015-03-20 09:05:01 -0700371 sessionId, transferType, uid, pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800372
Phil Burk33ff89b2015-11-30 11:16:01 -0800373 mThreadCanCallJava = threadCanCallJava;
jiabin156c6872017-10-06 09:47:15 -0700374 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800375 mSessionId = sessionId;
Phil Burk33ff89b2015-11-30 11:16:01 -0800376
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800377 switch (transferType) {
378 case TRANSFER_DEFAULT:
379 if (sharedBuffer != 0) {
380 transferType = TRANSFER_SHARED;
381 } else if (cbf == NULL || threadCanCallJava) {
382 transferType = TRANSFER_SYNC;
383 } else {
384 transferType = TRANSFER_CALLBACK;
385 }
386 break;
387 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700388 case TRANSFER_SYNC_NOTIF_CALLBACK:
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800389 if (cbf == NULL || sharedBuffer != 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700390 ALOGE("%s(): Transfer type %s but cbf == NULL || sharedBuffer != 0",
391 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800392 status = BAD_VALUE;
393 goto exit;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800394 }
395 break;
396 case TRANSFER_OBTAIN:
397 case TRANSFER_SYNC:
398 if (sharedBuffer != 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700399 ALOGE("%s(): Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800400 status = BAD_VALUE;
401 goto exit;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800402 }
403 break;
404 case TRANSFER_SHARED:
405 if (sharedBuffer == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700406 ALOGE("%s(): Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800407 status = BAD_VALUE;
408 goto exit;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800409 }
410 break;
411 default:
Andy Hungfb8ede22018-09-12 19:03:24 -0700412 ALOGE("%s(): Invalid transfer type %d",
413 __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800414 status = BAD_VALUE;
415 goto exit;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800416 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800417 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800418 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700419 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800420
Andy Hungfb8ede22018-09-12 19:03:24 -0700421 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700422 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800423
Andy Hungfb8ede22018-09-12 19:03:24 -0700424 ALOGV("%s(): streamType %d frameCount %zu flags %04x",
425 __func__, streamType, frameCount, flags);
Eric Laurent1a9ed112012-03-20 18:36:01 -0700426
Glenn Kasten53cec222013-08-29 09:01:02 -0700427 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700428 if (mAudioTrack != 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700429 ALOGE("%s(): Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800430 status = INVALID_OPERATION;
431 goto exit;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800432 }
433
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800434 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800435 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700436 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800437 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700438 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800439 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700440 ALOGE("%s(): Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800441 status = BAD_VALUE;
442 goto exit;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700443 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700444 mStreamType = streamType;
Eric Laurente83b55d2014-11-14 10:06:21 -0800445
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700446 } else {
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700447 // stream type shouldn't be looked at, this track has audio attributes
448 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
Andy Hungfb8ede22018-09-12 19:03:24 -0700449 ALOGV("%s(): Building AudioTrack with attributes:"
450 " usage=%d content=%d flags=0x%x tags=[%s]",
451 __func__,
452 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
Eric Laurente83b55d2014-11-14 10:06:21 -0800453 mStreamType = AUDIO_STREAM_DEFAULT;
François Gaffie58d4be52018-11-06 15:30:12 +0100454 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800455 }
Glenn Kastenea7939a2012-03-14 12:56:26 -0700456
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800457 // these below should probably come from the audioFlinger too...
Glenn Kastene1c39622012-01-04 09:36:37 -0800458 if (format == AUDIO_FORMAT_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700459 format = AUDIO_FORMAT_PCM_16_BIT;
Phil Burkfdb3c072016-02-09 10:47:02 -0800460 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
461 mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800462 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800463
464 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700465 if (!audio_is_valid_format(format)) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700466 ALOGE("%s(): Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800467 status = BAD_VALUE;
468 goto exit;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800469 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800470 mFormat = format;
Eric Laurentc2f1f072009-07-17 12:17:14 -0700471
Glenn Kasten8ba90322013-10-30 11:29:27 -0700472 if (!audio_is_output_channel(channelMask)) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700473 ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800474 status = BAD_VALUE;
475 goto exit;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700476 }
Glenn Kastene3247bf2014-02-24 15:19:07 -0800477 mChannelMask = channelMask;
Eric Laurentf32d7812017-11-30 14:44:07 -0800478 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800479 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700480
Eric Laurentc2f1f072009-07-17 12:17:14 -0700481 // force direct flag if format is not linear PCM
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100482 // or offload was requested
483 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
484 || !audio_is_linear_pcm(format)) {
485 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
Andy Hungfb8ede22018-09-12 19:03:24 -0700486 ? "%s(): Offload request, forcing to Direct Output"
487 : "%s(): Not linear PCM, forcing to Direct Output",
488 __func__);
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700489 flags = (audio_output_flags_t)
Glenn Kasten3acbd052012-02-28 10:39:56 -0800490 // FIXME why can't we allow direct AND fast?
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700491 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
Eric Laurentc2f1f072009-07-17 12:17:14 -0700492 }
493
Eric Laurentd1f69b02014-12-15 14:33:13 -0800494 // force direct flag if HW A/V sync requested
495 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
496 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
497 }
498
Glenn Kastenb7730382014-04-30 15:50:31 -0700499 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
Phil Burkfdb3c072016-02-09 10:47:02 -0800500 if (audio_has_proportional_frames(format)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700501 mFrameSize = channelCount * audio_bytes_per_sample(format);
502 } else {
503 mFrameSize = sizeof(uint8_t);
504 }
Glenn Kastene3aa6592012-12-04 12:22:46 -0800505 } else {
Phil Burkfdb3c072016-02-09 10:47:02 -0800506 ALOG_ASSERT(audio_has_proportional_frames(format));
Glenn Kastenb7730382014-04-30 15:50:31 -0700507 mFrameSize = channelCount * audio_bytes_per_sample(format);
Glenn Kastenb7730382014-04-30 15:50:31 -0700508 // createTrack will return an error if PCM format is not supported by server,
509 // so no need to check for specific PCM formats here
Glenn Kastene3aa6592012-12-04 12:22:46 -0800510 }
511
Eric Laurent0d6db582014-11-12 18:39:44 -0800512 // sampling rate must be specified for direct outputs
513 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Eric Laurentf32d7812017-11-30 14:44:07 -0800514 status = BAD_VALUE;
515 goto exit;
Eric Laurent0d6db582014-11-12 18:39:44 -0800516 }
517 mSampleRate = sampleRate;
Lajos Molnar3a474aa2015-04-24 17:10:07 -0700518 mOriginalSampleRate = sampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -0700519 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Andy Hungff874dc2016-04-11 16:49:09 -0700520 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
521 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800522
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800523 // Make copy of input parameter offloadInfo so that in the future:
524 // (a) createTrack_l doesn't need it as an input parameter
525 // (b) we can support re-creation of offloaded tracks
526 if (offloadInfo != NULL) {
527 mOffloadInfoCopy = *offloadInfo;
528 mOffloadInfo = &mOffloadInfoCopy;
529 } else {
530 mOffloadInfo = NULL;
Eric Laurent20b9ef02016-12-05 11:03:16 -0800531 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800532 }
533
Glenn Kasten66e46352014-01-16 17:44:23 -0800534 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
535 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800536 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800537 // mFrameCount is initialized in createTrack_l
Glenn Kastenb6037442012-11-14 13:42:25 -0800538 mReqFrameCount = frameCount;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700539 if (notificationFrames >= 0) {
540 mNotificationFramesReq = notificationFrames;
541 mNotificationsPerBufferReq = 0;
542 } else {
543 if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700544 ALOGE("%s(): notificationFrames=%d not permitted for non-fast track",
545 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800546 status = BAD_VALUE;
547 goto exit;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700548 }
549 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700550 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
551 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800552 status = BAD_VALUE;
553 goto exit;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700554 }
555 mNotificationFramesReq = 0;
556 const uint32_t minNotificationsPerBuffer = 1;
557 const uint32_t maxNotificationsPerBuffer = 8;
558 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
559 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
560 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700561 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
562 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700563 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
564 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800565 mNotificationFramesAct = 0;
Eric Laurentf32d7812017-11-30 14:44:07 -0800566 callingPid = IPCThreadState::self()->getCallingPid();
567 myPid = getpid();
568 if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800569 mClientUid = IPCThreadState::self()->getCallingUid();
570 } else {
571 mClientUid = uid;
572 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800573 if (pid == -1 || (callingPid != myPid)) {
574 mClientPid = callingPid;
Marco Nelissend457c972014-02-11 08:47:07 -0800575 } else {
576 mClientPid = pid;
577 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700578 mAuxEffectId = 0;
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -0800579 mOrigFlags = mFlags = flags;
Glenn Kasten4a4a0952012-03-19 11:38:14 -0700580 mCbf = cbf;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700581
Glenn Kastena997e7a2012-08-07 09:44:19 -0700582 if (cbf != NULL) {
Andy Hungca353672019-03-06 11:54:38 -0800583 mAudioTrackThread = new AudioTrackThread(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700584 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700585 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700586 }
587
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800588 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100589 {
590 AutoMutex lock(mLock);
591 status = createTrack_l();
592 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700593 if (status != NO_ERROR) {
594 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100595 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
596 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700597 mAudioTrackThread.clear();
598 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800599 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700600 }
601
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800602 mUserData = user;
Andy Hung4ede21d2014-12-12 15:37:34 -0800603 mLoopCount = 0;
604 mLoopStart = 0;
605 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800606 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800607 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700608 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800609 mNewPosition = 0;
610 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700611 mPosition = 0;
612 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700613 mStartNs = 0;
614 mStartFromZeroUs = 0;
Andy Hung8b0bfd92019-12-23 13:11:11 -0800615 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid, mClientUid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800616 mSequence = 1;
617 mObservedSequence = mSequence;
618 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700619 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700620 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700621 mTimestampRetrogradePositionReported = false;
622 mTimestampRetrogradeTimeReported = false;
623 mTimestampStallReported = false;
624 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700625 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700626 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800627 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800628 mFramesWritten = 0;
629 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700630 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700631 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800632
633exit:
634 mStatus = status;
635 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800636}
637
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800638// -------------------------------------------------------------------------
639
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100640status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800641{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800642 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800643
Andy Hung10fb4be2020-05-27 22:22:22 -0700644 if (mState == STATE_ACTIVE) {
645 return INVALID_OPERATION;
646 }
647
648 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
649
650 // Defer logging here due to OpenSL ES repeated start calls.
651 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
652 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800653 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700654 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800655 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700656 .set(AMEDIAMETRICS_PROP_CALLERNAME,
657 mCallerName.empty()
658 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
659 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800660 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700661 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800662 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
663 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
664 .record(); });
665
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800666
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800667 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800668
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800669 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100670 if (previousState == STATE_PAUSED_STOPPING) {
671 mState = STATE_STOPPING;
672 } else {
673 mState = STATE_ACTIVE;
674 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700675 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700676
677 // save start timestamp
678 if (isOffloadedOrDirect_l()) {
679 if (getTimestamp_l(mStartTs) != OK) {
680 mStartTs.mPosition = 0;
681 }
682 } else {
683 if (getTimestamp_l(&mStartEts) != OK) {
684 mStartEts.clear();
685 }
686 }
Andy Hungffa36952017-08-17 10:41:51 -0700687 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800688 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
689 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700690 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700691 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700692 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700693 mTimestampRetrogradePositionReported = false;
694 mTimestampRetrogradeTimeReported = false;
695 mTimestampStallReported = false;
696 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700697 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700698
Andy Hung65ffdfc2016-10-10 15:52:11 -0700699 if (!isOffloadedOrDirect_l()
700 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700701 // Server side has consumed something, but is it finished consuming?
702 // It is possible since flush and stop are asynchronous that the server
703 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700704 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800705 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700706 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700707 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
708 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700709 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700710 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
711 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700712 }
Andy Hunge1e98462016-04-12 10:18:51 -0700713 mFramesWritten = 0;
714 mProxy->clearTimestamp(); // need new server push for valid timestamp
715 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700716
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700717 // For offloaded tracks, we don't know if the hardware counters are really zero here,
718 // since the flush is asynchronous and stop may not fully drain.
719 // We save the time when the track is started to later verify whether
720 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700721 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700722
Eric Laurentec9a0322013-08-28 10:23:01 -0700723 // force refresh of remaining frames by processAudioBuffer() as last
724 // write before stop could be partial.
725 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900726
727 // for static track, clear the old flags when starting from stopped state
728 if (mSharedBuffer != 0) {
729 android_atomic_and(
730 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
731 &mCblk->mFlags);
732 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800733 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700734 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700735 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800736
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800737 if (!(flags & CBLK_INVALID)) {
738 status = mAudioTrack->start();
739 if (status == DEAD_OBJECT) {
740 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800741 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800742 }
743 if (flags & CBLK_INVALID) {
744 status = restoreTrack_l("start");
745 }
746
Andy Hung79629f02016-03-24 13:57:40 -0700747 // resume or pause the callback thread as needed.
748 sp<AudioTrackThread> t = mAudioTrackThread;
749 if (status == NO_ERROR) {
750 if (t != 0) {
751 if (previousState == STATE_STOPPING) {
752 mProxy->interrupt();
753 } else {
754 t->resume();
755 }
756 } else {
757 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
758 get_sched_policy(0, &mPreviousSchedulingGroup);
759 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
760 }
Andy Hung39399b62017-04-21 15:07:45 -0700761
762 // Start our local VolumeHandler for restoration purposes.
763 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -0700764 } else {
Eric Laurent973db022018-11-20 14:54:31 -0800765 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800766 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800767 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100768 if (previousState != STATE_STOPPING) {
769 t->pause();
770 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800771 } else {
Glenn Kasten87913512011-06-22 16:15:25 -0700772 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -0700773 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800774 }
775 }
776
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100777 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800778}
779
780void AudioTrack::stop()
781{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800782 const int64_t beginNs = systemTime();
783
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800784 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700785 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800786 mediametrics::LogItem(mMetricsId)
787 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -0700788 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800789 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -0700790 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
791 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -0700792 .record();
Phil Burka9876702020-04-20 18:16:15 -0700793 });
Andy Hungb68f5eb2019-12-03 16:49:17 -0800794
Eric Laurent973db022018-11-20 14:54:31 -0800795 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700796
Glenn Kasten397edb32013-08-30 15:10:13 -0700797 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800798 return;
799 }
800
Glenn Kasten23a75452014-01-13 10:37:17 -0800801 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100802 mState = STATE_STOPPING;
803 } else {
804 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -0800805 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -0800806 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -0700807 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100808 }
809
Andy Hung1d3556d2018-03-29 16:30:14 -0700810 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800811 mProxy->interrupt();
812 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -0700813
814 // Note: legacy handling - stop does not clear playback marker
815 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -0800816
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800817 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -0800818 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -0800819 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
820 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800821 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100822
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800823 sp<AudioTrackThread> t = mAudioTrackThread;
824 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -0800825 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100826 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -0800827 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -0800828 // causes wake up of the playback thread, that will callback the client for
829 // EVENT_STREAM_END in processAudioBuffer()
830 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100831 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800832 } else {
833 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
834 set_sched_policy(0, mPreviousSchedulingGroup);
835 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800836}
837
838bool AudioTrack::stopped() const
839{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -0800840 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800841 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800842}
843
844void AudioTrack::flush()
845{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800846 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -0700847 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700848 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800849 mediametrics::LogItem(mMetricsId)
850 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -0700851 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800852 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
853 .record(); });
854
Eric Laurent973db022018-11-20 14:54:31 -0800855 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700856
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800857 if (mSharedBuffer != 0) {
858 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -0800859 }
Andy Hung4c5ed302018-05-09 11:16:21 -0700860 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800861 return;
862 }
863 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -0800864}
865
Eric Laurent1703cdf2011-03-07 14:52:59 -0800866void AudioTrack::flush_l()
867{
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800868 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -0700869
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700870 // clear playback marker and periodic update counter
871 mMarkerPosition = 0;
872 mMarkerReached = false;
873 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100874 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -0700875
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800876 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -0700877 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -0800878 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100879 mProxy->interrupt();
880 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800881 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -0800882 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800883}
884
885void AudioTrack::pause()
886{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800887 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -0800888 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -0700889 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800890 mediametrics::LogItem(mMetricsId)
891 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -0700892 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800893 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
894 .record(); });
895
Eric Laurent973db022018-11-20 14:54:31 -0800896 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700897
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100898 if (mState == STATE_ACTIVE) {
899 mState = STATE_PAUSED;
900 } else if (mState == STATE_STOPPING) {
901 mState = STATE_PAUSED_STOPPING;
902 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800903 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800904 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800905 mProxy->interrupt();
906 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800907
Marco Nelissen3a90f282014-03-10 11:21:43 -0700908 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -0700909 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700910 // An offload output can be re-used between two audio tracks having
911 // the same configuration. A timestamp query for a paused track
912 // while the other is running would return an incorrect time.
913 // To fix this, cache the playback position on a pause() and return
914 // this time when requested until the track is resumed.
915
916 // OffloadThread sends HAL pause in its threadLoop. Time saved
917 // here can be slightly off.
918
919 // TODO: check return code for getRenderPosition.
920
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800921 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800922 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -0700923 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -0800924 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800925 }
926 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800927}
928
Eric Laurentbe916aa2010-06-01 23:49:17 -0700929status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800930{
Glenn Kastenc56f3422014-03-21 17:53:17 -0700931 // This duplicates a test by AudioTrack JNI, but that is not the only caller
932 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
933 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -0700934 return BAD_VALUE;
935 }
936
Andy Hungb68f5eb2019-12-03 16:49:17 -0800937 mediametrics::LogItem(mMetricsId)
938 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
939 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
940 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
941 .record();
942
Eric Laurent1703cdf2011-03-07 14:52:59 -0800943 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -0800944 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
945 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800946
Glenn Kastenc56f3422014-03-21 17:53:17 -0700947 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -0700948
Glenn Kasten23a75452014-01-13 10:37:17 -0800949 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -0700950 mAudioTrack->signal();
951 }
Eric Laurentbe916aa2010-06-01 23:49:17 -0700952 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800953}
954
Glenn Kastenb1c09932012-02-27 16:21:04 -0800955status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800956{
Glenn Kastenb1c09932012-02-27 16:21:04 -0800957 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -0700958}
959
Eric Laurent2beeb502010-07-16 07:43:46 -0700960status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -0700961{
Glenn Kastenc56f3422014-03-21 17:53:17 -0700962 // This duplicates a test by AudioTrack JNI, but that is not the only caller
963 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -0700964 return BAD_VALUE;
965 }
966
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800967 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -0700968 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -0800969 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -0700970
971 return NO_ERROR;
972}
973
Glenn Kastena5224f32012-01-04 12:41:44 -0800974void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -0700975{
976 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800977 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700978 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800979}
980
Glenn Kasten3b16c762012-11-14 08:44:39 -0800981status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800982{
Andy Hung5cbb5782015-03-27 18:39:59 -0700983 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -0800984 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -0700985
Andy Hung5cbb5782015-03-27 18:39:59 -0700986 if (rate == mSampleRate) {
987 return NO_ERROR;
988 }
jiabinf4de6112018-12-19 12:40:08 -0800989 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
990 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -0800991 return INVALID_OPERATION;
992 }
Eric Laurent0d6db582014-11-12 18:39:44 -0800993 if (mOutput == AUDIO_IO_HANDLE_NONE) {
994 return NO_INIT;
995 }
Andy Hung5cbb5782015-03-27 18:39:59 -0700996 // NOTE: it is theoretically possible, but highly unlikely, that a device change
997 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800998 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -0800999 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001000 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001001 }
Andy Hung26145642015-04-15 21:56:53 -07001002 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001003 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001004 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001005 return BAD_VALUE;
1006 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001007 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001008
Glenn Kastene3aa6592012-12-04 12:22:46 -08001009 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001010 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001011
Eric Laurent57326622009-07-07 07:10:45 -07001012 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001013}
1014
Glenn Kastena5224f32012-01-04 12:41:44 -08001015uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001016{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001017 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001018
1019 // sample rate can be updated during playback by the offloaded decoder so we need to
1020 // query the HAL and update if needed.
1021// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001022 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001023 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001024 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001025 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001026 if (status == NO_ERROR) {
1027 mSampleRate = sampleRate;
1028 }
1029 }
1030 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001031 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001032}
1033
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001034uint32_t AudioTrack::getOriginalSampleRate() const
1035{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001036 return mOriginalSampleRate;
1037}
1038
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001039status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001040{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001041 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001042 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001043 return NO_ERROR;
1044 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001045 if (isOffloadedOrDirect_l()) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001046 return INVALID_OPERATION;
1047 }
1048 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1049 return INVALID_OPERATION;
1050 }
Andy Hungff874dc2016-04-11 16:49:09 -07001051
Andy Hungfb8ede22018-09-12 19:03:24 -07001052 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001053 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001054 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001055 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1056 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1057 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001058 AudioPlaybackRate playbackRateTemp = playbackRate;
1059 playbackRateTemp.mSpeed = effectiveSpeed;
1060 playbackRateTemp.mPitch = effectivePitch;
1061
Andy Hungfb8ede22018-09-12 19:03:24 -07001062 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001063 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001064
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001065 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001066 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001067 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001068 return BAD_VALUE;
1069 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001070 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001071 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001072 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001073 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001074 return BAD_VALUE;
1075 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001076
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001077 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001078 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1079 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001080 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001081 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001082 return BAD_VALUE;
1083 }
1084
Dan Austine34eae22015-10-27 16:14:52 -07001085 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001086 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001087 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001088 return BAD_VALUE;
1089 }
1090 mPlaybackRate = playbackRate;
1091 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001092 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001093 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001094
1095 mediametrics::LogItem(mMetricsId)
1096 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1097 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1098 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1099 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1100 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1101 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1102 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1103 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1104 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1105 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1106 .record();
1107
Andy Hung8edb8dc2015-03-26 19:13:55 -07001108 return NO_ERROR;
1109}
1110
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001111const AudioPlaybackRate& AudioTrack::getPlaybackRate() const
Andy Hung8edb8dc2015-03-26 19:13:55 -07001112{
1113 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001114 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001115}
1116
Phil Burkc0adecb2016-01-08 12:44:11 -08001117ssize_t AudioTrack::getBufferSizeInFrames()
1118{
1119 AutoMutex lock(mLock);
1120 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1121 return NO_INIT;
1122 }
Phil Burka9876702020-04-20 18:16:15 -07001123
Phil Burke8972b02016-03-04 11:29:57 -08001124 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001125}
1126
Andy Hungf2c87b32016-04-07 19:49:29 -07001127status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1128{
1129 if (duration == nullptr) {
1130 return BAD_VALUE;
1131 }
1132 AutoMutex lock(mLock);
1133 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1134 return NO_INIT;
1135 }
1136 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1137 if (bufferSizeInFrames < 0) {
1138 return (status_t)bufferSizeInFrames;
1139 }
1140 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1141 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1142 return NO_ERROR;
1143}
1144
Phil Burkc0adecb2016-01-08 12:44:11 -08001145ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1146{
1147 AutoMutex lock(mLock);
1148 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1149 return NO_INIT;
1150 }
1151 // Reject if timed track or compressed audio.
Glenn Kastend79072e2016-01-06 08:41:20 -08001152 if (!audio_is_linear_pcm(mFormat)) {
Phil Burkc0adecb2016-01-08 12:44:11 -08001153 return INVALID_OPERATION;
1154 }
Phil Burka9876702020-04-20 18:16:15 -07001155
1156 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1157 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1158 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001159 android::mediametrics::LogItem(mMetricsId)
1160 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1161 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1162 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1163 .record();
Phil Burka9876702020-04-20 18:16:15 -07001164 }
1165 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001166}
1167
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001168status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1169{
Glenn Kastend79072e2016-01-06 08:41:20 -08001170 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001171 return INVALID_OPERATION;
1172 }
1173
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001174 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001175 ;
1176 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1177 loopEnd - loopStart >= MIN_LOOP) {
1178 ;
1179 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001180 return BAD_VALUE;
1181 }
1182
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001183 AutoMutex lock(mLock);
1184 // See setPosition() regarding setting parameters such as loop points or position while active
1185 if (mState == STATE_ACTIVE) {
1186 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001187 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001188 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001189 return NO_ERROR;
1190}
1191
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001192void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1193{
Andy Hung4ede21d2014-12-12 15:37:34 -08001194 // We do not update the periodic notification point.
1195 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1196 mLoopCount = loopCount;
1197 mLoopEnd = loopEnd;
1198 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001199 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001200 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001201
1202 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001203}
1204
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001205status_t AudioTrack::setMarkerPosition(uint32_t marker)
1206{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001207 // The only purpose of setting marker position is to get a callback
Eric Laurentab5cdba2014-06-09 17:22:27 -07001208 if (mCbf == NULL || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001209 return INVALID_OPERATION;
1210 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001211
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001212 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001213 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001214 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001215
Andy Hung3c09c782014-12-29 18:39:32 -08001216 sp<AudioTrackThread> t = mAudioTrackThread;
1217 if (t != 0) {
1218 t->wake();
1219 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001220 return NO_ERROR;
1221}
1222
Glenn Kastena5224f32012-01-04 12:41:44 -08001223status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001224{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001225 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001226 return INVALID_OPERATION;
1227 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001228 if (marker == NULL) {
1229 return BAD_VALUE;
1230 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001231
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001232 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001233 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001234
1235 return NO_ERROR;
1236}
1237
1238status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1239{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001240 // The only purpose of setting position update period is to get a callback
Eric Laurentab5cdba2014-06-09 17:22:27 -07001241 if (mCbf == NULL || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001242 return INVALID_OPERATION;
1243 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001244
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001245 AutoMutex lock(mLock);
Glenn Kasten200092b2014-08-15 15:13:30 -07001246 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001247 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001248
Andy Hung3c09c782014-12-29 18:39:32 -08001249 sp<AudioTrackThread> t = mAudioTrackThread;
1250 if (t != 0) {
1251 t->wake();
1252 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001253 return NO_ERROR;
1254}
1255
Glenn Kastena5224f32012-01-04 12:41:44 -08001256status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001257{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001258 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001259 return INVALID_OPERATION;
1260 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001261 if (updatePeriod == NULL) {
1262 return BAD_VALUE;
1263 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001264
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001265 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001266 *updatePeriod = mUpdatePeriod;
1267
1268 return NO_ERROR;
1269}
1270
1271status_t AudioTrack::setPosition(uint32_t position)
1272{
Glenn Kastend79072e2016-01-06 08:41:20 -08001273 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001274 return INVALID_OPERATION;
1275 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001276 if (position > mFrameCount) {
1277 return BAD_VALUE;
1278 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001279
Eric Laurent1703cdf2011-03-07 14:52:59 -08001280 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001281 // Currently we require that the player is inactive before setting parameters such as position
1282 // or loop points. Otherwise, there could be a race condition: the application could read the
1283 // current position, compute a new position or loop parameters, and then set that position or
1284 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1285 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1286 // to specify how it wants to handle such scenarios.
1287 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001288 return INVALID_OPERATION;
1289 }
Andy Hung9b461582014-12-01 17:56:29 -08001290 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001291 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001292 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001293
1294 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001295 return NO_ERROR;
1296}
1297
Glenn Kasten200092b2014-08-15 15:13:30 -07001298status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001299{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001300 if (position == NULL) {
1301 return BAD_VALUE;
1302 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001303
Eric Laurent1703cdf2011-03-07 14:52:59 -08001304 AutoMutex lock(mLock);
Andy Hung7a490e72016-03-23 15:58:10 -07001305 // FIXME: offloaded and direct tracks call into the HAL for render positions
1306 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1307 // as we do not know the capability of the HAL for pcm position support and standby.
1308 // There may be some latency differences between the HAL position and the proxy position.
1309 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001310 uint32_t dspFrames = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001311
Eric Laurentab5cdba2014-06-09 17:22:27 -07001312 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001313 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001314 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001315 *position = mPausedPosition;
1316 return NO_ERROR;
1317 }
1318
Glenn Kasten142f5192014-03-25 17:44:59 -07001319 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001320 uint32_t halFrames; // actually unused
1321 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
1322 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001323 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001324 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
1325 // due to hardware latency. We leave this behavior for now.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001326 *position = dspFrames;
1327 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001328 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001329 (void) restoreTrack_l("getPosition");
1330 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1331 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001332 }
1333
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001334 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
Glenn Kasten200092b2014-08-15 15:13:30 -07001335 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
Andy Hung90e8a972015-11-09 16:42:40 -08001336 0 : updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001337 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001338 return NO_ERROR;
1339}
1340
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001341status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001342{
Glenn Kastend79072e2016-01-06 08:41:20 -08001343 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001344 return INVALID_OPERATION;
1345 }
1346 if (position == NULL) {
1347 return BAD_VALUE;
1348 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001349
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001350 AutoMutex lock(mLock);
1351 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001352 return NO_ERROR;
1353}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001354
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001355status_t AudioTrack::reload()
1356{
Glenn Kastend79072e2016-01-06 08:41:20 -08001357 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001358 return INVALID_OPERATION;
1359 }
1360
Eric Laurent1703cdf2011-03-07 14:52:59 -08001361 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001362 // See setPosition() regarding setting parameters such as loop points or position while active
1363 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001364 return INVALID_OPERATION;
1365 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001366 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001367 (void) updateAndGetPosition_l();
1368 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001369 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001370#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001371 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001372 // of loop count. Historically we have not restored loop count, start, end,
1373 // but it makes sense if one desires to repeat playing a particular sound.
1374 if (mLoopCount != 0) {
1375 mLoopCountNotified = mLoopCount;
1376 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1377 }
1378#endif
Andy Hung9b461582014-12-01 17:56:29 -08001379 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001380 return NO_ERROR;
1381}
1382
Glenn Kasten38e905b2014-01-13 10:21:48 -08001383audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001384{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001385 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001386 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001387}
1388
Paul McLeanaa981192015-03-21 09:55:15 -07001389status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
1390 AutoMutex lock(mLock);
1391 if (mSelectedDeviceId != deviceId) {
1392 mSelectedDeviceId = deviceId;
Eric Laurentfb00fc72017-05-25 18:17:12 -07001393 if (mStatus == NO_ERROR) {
1394 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
jiabin156c6872017-10-06 09:47:15 -07001395 mProxy->interrupt();
Eric Laurentfb00fc72017-05-25 18:17:12 -07001396 }
Paul McLeanaa981192015-03-21 09:55:15 -07001397 }
Eric Laurent493404d2015-04-21 15:07:36 -07001398 return NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001399}
1400
1401audio_port_handle_t AudioTrack::getOutputDevice() {
1402 AutoMutex lock(mLock);
1403 return mSelectedDeviceId;
1404}
1405
Eric Laurentad2e7b92017-09-14 20:06:42 -07001406// must be called with mLock held
1407void AudioTrack::updateRoutedDeviceId_l()
1408{
1409 // if the track is inactive, do not update actual device as the output stream maybe routed
1410 // to a device not relevant to this client because of other active use cases.
1411 if (mState != STATE_ACTIVE) {
1412 return;
1413 }
1414 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1415 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1416 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1417 mRoutedDeviceId = deviceId;
1418 }
1419 }
1420}
1421
Eric Laurent296fb132015-05-01 11:38:42 -07001422audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1423 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001424 updateRoutedDeviceId_l();
1425 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001426}
1427
Eric Laurentbe916aa2010-06-01 23:49:17 -07001428status_t AudioTrack::attachAuxEffect(int effectId)
1429{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001430 AutoMutex lock(mLock);
Eric Laurent2beeb502010-07-16 07:43:46 -07001431 status_t status = mAudioTrack->attachAuxEffect(effectId);
1432 if (status == NO_ERROR) {
1433 mAuxEffectId = effectId;
1434 }
1435 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001436}
1437
Eric Laurente83b55d2014-11-14 10:06:21 -08001438audio_stream_type_t AudioTrack::streamType() const
1439{
1440 if (mStreamType == AUDIO_STREAM_DEFAULT) {
François Gaffie58d4be52018-11-06 15:30:12 +01001441 return AudioSystem::attributesToStreamType(mAttributes);
Eric Laurente83b55d2014-11-14 10:06:21 -08001442 }
1443 return mStreamType;
1444}
1445
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001446uint32_t AudioTrack::latency()
1447{
1448 AutoMutex lock(mLock);
1449 updateLatency_l();
1450 return mLatency;
1451}
1452
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001453// -------------------------------------------------------------------------
1454
Eric Laurent1703cdf2011-03-07 14:52:59 -08001455// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001456void AudioTrack::updateLatency_l()
1457{
1458 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1459 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001460 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001461 } else {
1462 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001463 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001464 }
1465}
1466
Phil Burkadbb75a2017-06-16 12:19:42 -07001467// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1468#define MEDIA_CASE_ENUM(name) case name: return #name
1469const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1470 switch (transferType) {
1471 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1472 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1473 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1474 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1475 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001476 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001477 default:
1478 return "UNRECOGNIZED";
1479 }
1480}
1481
Glenn Kasten200092b2014-08-15 15:13:30 -07001482status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001483{
Eric Laurentf32d7812017-11-30 14:44:07 -08001484 status_t status;
1485 bool callbackAdded = false;
1486
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001487 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1488 if (audioFlinger == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001489 ALOGE("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001490 __func__, mPortId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001491 status = NO_INIT;
1492 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001493 }
1494
Eric Laurent21da6472017-11-09 16:29:26 -08001495 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001496 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1497 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001498 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001499 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001500 // either of these use cases:
1501 // use case 1: shared buffer
1502 bool sharedBuffer = mSharedBuffer != 0;
1503 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001504 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001505 (mTransfer == TRANSFER_CALLBACK) ||
1506 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001507 (mTransfer == TRANSFER_OBTAIN) ||
1508 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001509 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1510 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001511
Eric Laurent21da6472017-11-09 16:29:26 -08001512 bool fastAllowed = sharedBuffer || transferAllowed;
1513 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001514 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1515 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001516 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001517 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001518 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1519 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001520 }
1521
Eric Laurent21da6472017-11-09 16:29:26 -08001522 IAudioFlinger::CreateTrackInput input;
1523 if (mStreamType != AUDIO_STREAM_DEFAULT) {
François Gaffie58d4be52018-11-06 15:30:12 +01001524 input.attr = AudioSystem::streamTypeToAttributes(mStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001525 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001526 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001527 }
Eric Laurent21da6472017-11-09 16:29:26 -08001528 input.config = AUDIO_CONFIG_INITIALIZER;
1529 input.config.sample_rate = mSampleRate;
1530 input.config.channel_mask = mChannelMask;
1531 input.config.format = mFormat;
1532 input.config.offload_info = mOffloadInfoCopy;
1533 input.clientInfo.clientUid = mClientUid;
1534 input.clientInfo.clientPid = mClientPid;
1535 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001536 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001537 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1538 // application-level code follows all non-blocking design rules, the language runtime
1539 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001540 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001541 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001542 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001543 }
Eric Laurent21da6472017-11-09 16:29:26 -08001544 input.sharedBuffer = mSharedBuffer;
1545 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1546 input.speed = 1.0;
1547 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1548 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1549 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1550 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1551 }
1552 input.flags = mFlags;
1553 input.frameCount = mReqFrameCount;
1554 input.notificationFrameCount = mNotificationFramesReq;
1555 input.selectedDeviceId = mSelectedDeviceId;
1556 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001557 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001558
Eric Laurent21da6472017-11-09 16:29:26 -08001559 IAudioFlinger::CreateTrackOutput output;
1560
1561 sp<IAudioTrack> track = audioFlinger->createTrack(input,
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001562 output,
Eric Laurent21da6472017-11-09 16:29:26 -08001563 &status);
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001564
Eric Laurent21da6472017-11-09 16:29:26 -08001565 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001566 ALOGE("%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001567 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001568 if (status == NO_ERROR) {
1569 status = NO_INIT;
1570 }
1571 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001572 }
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001573 ALOG_ASSERT(track != 0);
1574
Eric Laurent21da6472017-11-09 16:29:26 -08001575 mFrameCount = output.frameCount;
1576 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1577 mRoutedDeviceId = output.selectedDeviceId;
1578 mSessionId = output.sessionId;
1579
1580 mSampleRate = output.sampleRate;
1581 if (mOriginalSampleRate == 0) {
1582 mOriginalSampleRate = mSampleRate;
1583 }
1584
1585 mAfFrameCount = output.afFrameCount;
1586 mAfSampleRate = output.afSampleRate;
1587 mAfLatency = output.afLatencyMs;
1588
1589 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1590
Glenn Kasten38e905b2014-01-13 10:21:48 -08001591 // AudioFlinger now owns the reference to the I/O handle,
1592 // so we are no longer responsible for releasing it.
1593
Glenn Kasten7fd04222016-02-02 12:38:16 -08001594 // FIXME compare to AudioRecord
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001595 sp<IMemory> iMem = track->getCblk();
1596 if (iMem == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08001597 ALOGE("%s(%d): Could not get control block", __func__, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001598 status = NO_INIT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001599 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001600 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001601 // TODO: Using unsecurePointer() has some associated security pitfalls
1602 // (see declaration for details).
1603 // Either document why it is safe in this case or address the
1604 // issue (e.g. by copying).
1605 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08001606 if (iMemPointer == NULL) {
Eric Laurent973db022018-11-20 14:54:31 -08001607 ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001608 status = NO_INIT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001609 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08001610 }
Glenn Kasten53cec222013-08-29 09:01:02 -07001611 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001612 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08001613 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001614 mDeathNotifier.clear();
1615 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001616 mAudioTrack = track;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001617 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07001618 IPCThreadState::self()->flushCommands();
1619
Glenn Kasten0cde0762014-01-16 15:06:36 -08001620 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001621 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08001622
Glenn Kastena07f17c2013-04-23 12:39:37 -07001623 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08001624 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08001625 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001626 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001627 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08001628 if (!mThreadCanCallJava) {
1629 mAwaitBoost = true;
1630 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001631 } else {
Phil Burkcc6ed2d2020-05-18 13:06:54 -07001632 ALOGD("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001633 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001634 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001635 }
Eric Laurent21da6472017-11-09 16:29:26 -08001636 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001637
Eric Laurentad2e7b92017-09-14 20:06:42 -07001638 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07001639 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07001640 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07001641 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001642 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07001643 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001644 callbackAdded = true;
1645 }
1646
Eric Laurent09f1ed22019-04-24 17:45:17 -07001647 mPortId = output.portId;
Glenn Kasten38e905b2014-01-13 10:21:48 -08001648 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08001649 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001650 mRefreshRemaining = true;
1651
1652 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
1653 // is the value of pointer() for the shared buffer, otherwise buffers points
1654 // immediately after the control block. This address is for the mapping within client
1655 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
1656 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08001657 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07001658 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001659 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001660 // TODO: Using unsecurePointer() has some associated security pitfalls
1661 // (see declaration for details).
1662 // Either document why it is safe in this case or address the
1663 // issue (e.g. by copying).
1664 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07001665 if (buffers == NULL) {
Eric Laurent973db022018-11-20 14:54:31 -08001666 ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001667 status = NO_INIT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001668 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07001669 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001670 }
1671
Eric Laurent2beeb502010-07-16 07:43:46 -07001672 mAudioTrack->attachAuxEffect(mAuxEffectId);
Glenn Kasten5f631512014-02-24 15:16:07 -08001673
Glenn Kasten093000f2012-05-03 09:35:36 -07001674 // If IAudioTrack is re-created, don't let the requested frameCount
1675 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08001676 if (mFrameCount > mReqFrameCount) {
1677 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07001678 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001679
Andy Hungd7bd69e2015-07-24 07:52:41 -07001680 // reset server position to 0 as we have new cblk.
1681 mServer = 0;
1682
Glenn Kastene3aa6592012-12-04 12:22:46 -08001683 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08001684 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001685 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08001686 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001687 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001688 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001689 mProxy = mStaticProxy;
1690 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09001691
1692 mProxy->setVolumeLR(gain_minifloat_pack(
1693 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
1694 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
1695
Glenn Kastene3aa6592012-12-04 12:22:46 -08001696 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001697 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
1698 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
1699 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001700 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001701
1702 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
1703 playbackRateTemp.mSpeed = effectiveSpeed;
1704 playbackRateTemp.mPitch = effectivePitch;
1705 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001706 mProxy->setMinimum(mNotificationFramesAct);
1707
1708 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08001709 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001710
Andy Hungb68f5eb2019-12-03 16:49:17 -08001711 // This is the first log sent from the AudioTrack client.
1712 // The creation of the audio track by AudioFlinger (in the code above)
1713 // is the first log of the AudioTrack and must be present before
1714 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07001715
Andy Hungb68f5eb2019-12-03 16:49:17 -08001716 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
1717 mediametrics::LogItem(mMetricsId)
1718 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
1719 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07001720 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
1721 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08001722 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
1723 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08001724 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
1725 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
1726 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
1727 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
1728 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
1729 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
1730 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
1731 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
1732 // the following are NOT immutable
1733 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
1734 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
1735 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1736 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
1737 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1738 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1739 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1740 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1741 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
1742 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1743 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
1744 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1745 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
1746 .record();
1747
1748 // mSendLevel
1749 // mReqFrameCount?
1750 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
1751 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
1752
Glenn Kasten38e905b2014-01-13 10:21:48 -08001753 }
1754
Eric Laurentf32d7812017-11-30 14:44:07 -08001755exit:
1756 if (status != NO_ERROR && callbackAdded) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07001757 // note: mOutput is always valid is callbackAdded is true
Eric Laurent09f1ed22019-04-24 17:45:17 -07001758 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001759 }
Eric Laurentf32d7812017-11-30 14:44:07 -08001760
1761 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08001762
1763 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08001764 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001765}
1766
Glenn Kastenb46f3942015-03-09 12:00:30 -07001767status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001768{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001769 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07001770 if (nonContig != NULL) {
1771 *nonContig = 0;
1772 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001773 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07001774 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001775 if (mTransfer != TRANSFER_OBTAIN) {
1776 audioBuffer->frameCount = 0;
1777 audioBuffer->size = 0;
1778 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07001779 if (nonContig != NULL) {
1780 *nonContig = 0;
1781 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001782 return INVALID_OPERATION;
1783 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07001784
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001785 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08001786 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001787 if (waitCount == -1) {
1788 requested = &ClientProxy::kForever;
1789 } else if (waitCount == 0) {
1790 requested = &ClientProxy::kNonBlocking;
1791 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07001792 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001793 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07001794 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001795 requested = &timeout;
1796 } else {
Eric Laurent973db022018-11-20 14:54:31 -08001797 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001798 requested = NULL;
1799 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07001800 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001801}
Eric Laurent1703cdf2011-03-07 14:52:59 -08001802
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001803status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
1804 struct timespec *elapsed, size_t *nonContig)
1805{
1806 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
1807 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001808
1809 Proxy::Buffer buffer;
1810 status_t status = NO_ERROR;
1811
1812 static const int32_t kMaxTries = 5;
1813 int32_t tryCounter = kMaxTries;
1814
1815 do {
1816 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
1817 // keep them from going away if another thread re-creates the track during obtainBuffer()
1818 sp<AudioTrackClientProxy> proxy;
1819 sp<IMemory> iMem;
1820
1821 { // start of lock scope
1822 AutoMutex lock(mLock);
1823
Glenn Kasten305996c2020-01-27 08:03:37 -08001824 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001825 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
1826 if (status == DEAD_OBJECT) {
1827 // re-create track, unless someone else has already done so
1828 if (newSequence == oldSequence) {
1829 status = restoreTrack_l("obtainBuffer");
1830 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001831 buffer.mFrameCount = 0;
1832 buffer.mRaw = NULL;
1833 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001834 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001835 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001836 }
1837 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001838 oldSequence = newSequence;
1839
Eric Laurent4d231dc2016-03-11 18:38:23 -08001840 if (status == NOT_ENOUGH_DATA) {
1841 restartIfDisabled();
1842 }
1843
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001844 // Keep the extra references
1845 proxy = mProxy;
1846 iMem = mCblkMemory;
1847
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001848 if (mState == STATE_STOPPING) {
1849 status = -EINTR;
1850 buffer.mFrameCount = 0;
1851 buffer.mRaw = NULL;
1852 buffer.mNonContig = 0;
1853 break;
1854 }
1855
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001856 // Non-blocking if track is stopped or paused
1857 if (mState != STATE_ACTIVE) {
1858 requested = &ClientProxy::kNonBlocking;
1859 }
1860
1861 } // end of lock scope
1862
1863 buffer.mFrameCount = audioBuffer->frameCount;
1864 // FIXME starts the requested timeout and elapsed over from scratch
1865 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08001866 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001867
1868 audioBuffer->frameCount = buffer.mFrameCount;
Andy Hungabdb9902015-01-12 15:08:22 -08001869 audioBuffer->size = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001870 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08001871 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001872 if (nonContig != NULL) {
1873 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001874 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001875 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001876}
1877
Glenn Kasten54a8a452015-03-09 12:03:00 -07001878void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001879{
Glenn Kasten3f02be22015-03-09 11:59:04 -07001880 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001881 if (mTransfer == TRANSFER_SHARED) {
1882 return;
1883 }
1884
Andy Hungabdb9902015-01-12 15:08:22 -08001885 size_t stepCount = audioBuffer->size / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001886 if (stepCount == 0) {
1887 return;
1888 }
1889
1890 Proxy::Buffer buffer;
1891 buffer.mFrameCount = stepCount;
1892 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001893
Eric Laurent1703cdf2011-03-07 14:52:59 -08001894 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08001895 if (audioBuffer->sequence != mSequence) {
1896 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
1897 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
1898 __func__, audioBuffer->sequence, mSequence);
1899 return;
1900 }
Glenn Kasten200092b2014-08-15 15:13:30 -07001901 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001902 mInUnderrun = false;
1903 mProxy->releaseBuffer(&buffer);
1904
1905 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08001906 restartIfDisabled();
1907}
1908
1909void AudioTrack::restartIfDisabled()
1910{
1911 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
1912 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001913 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08001914 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08001915 // FIXME ignoring status
1916 mAudioTrack->start();
Eric Laurentdf839842012-05-31 14:27:14 -07001917 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001918}
1919
1920// -------------------------------------------------------------------------
1921
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08001922ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001923{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001924 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001925 return INVALID_OPERATION;
1926 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001927
Eric Laurentab5cdba2014-06-09 17:22:27 -07001928 if (isDirect()) {
1929 AutoMutex lock(mLock);
1930 int32_t flags = android_atomic_and(
1931 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
1932 &mCblk->mFlags);
1933 if (flags & CBLK_INVALID) {
1934 return DEAD_OBJECT;
1935 }
1936 }
1937
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001938 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00001939 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08001940 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07001941 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08001942 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001943 return BAD_VALUE;
1944 }
1945
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001946 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001947 Buffer audioBuffer;
1948
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001949 while (userSize >= mFrameSize) {
1950 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001951
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08001952 status_t err = obtainBuffer(&audioBuffer,
1953 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001954 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001955 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001956 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07001957 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07001958 if (err == TIMED_OUT || err == -EINTR) {
1959 err = WOULD_BLOCK;
1960 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001961 return ssize_t(err);
1962 }
1963
Glenn Kastenae4b8792015-03-20 09:04:21 -07001964 size_t toWrite = audioBuffer.size;
Andy Hungabdb9902015-01-12 15:08:22 -08001965 memcpy(audioBuffer.i8, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001966 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001967 userSize -= toWrite;
1968 written += toWrite;
1969
1970 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001971 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001972
Andy Hungea2b9c02016-02-12 17:06:53 -08001973 if (written > 0) {
1974 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001975
1976 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
1977 const sp<AudioTrackThread> t = mAudioTrackThread;
1978 if (t != 0) {
1979 // causes wake up of the playback thread, that will callback the client for
1980 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
1981 t->wake();
1982 }
1983 }
Andy Hungea2b9c02016-02-12 17:06:53 -08001984 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001985
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001986 return written;
1987}
1988
1989// -------------------------------------------------------------------------
1990
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08001991nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001992{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001993 // Currently the AudioTrack thread is not created if there are no callbacks.
1994 // Would it ever make sense to run the thread, even without callbacks?
1995 // If so, then replace this by checks at each use for mCbf != NULL.
1996 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
1997
Eric Laurent1703cdf2011-03-07 14:52:59 -08001998 mLock.lock();
Glenn Kastena07f17c2013-04-23 12:39:37 -07001999 if (mAwaitBoost) {
2000 mAwaitBoost = false;
2001 mLock.unlock();
2002 static const int32_t kMaxTries = 5;
2003 int32_t tryCounter = kMaxTries;
2004 uint32_t pollUs = 10000;
2005 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002006 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002007 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2008 break;
2009 }
2010 usleep(pollUs);
2011 pollUs <<= 1;
2012 } while (tryCounter-- > 0);
2013 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002014 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002015 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002016 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002017 // Run again immediately
2018 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002019 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002020
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002021 // Can only reference mCblk while locked
2022 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002023 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002024
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002025 // Check for track invalidation
2026 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002027 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2028 // AudioSystem cache. We should not exit here but after calling the callback so
2029 // that the upper layers can recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002030 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002031 status_t status __unused = restoreTrack_l("processAudioBuffer");
2032 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002033 // after restoration, continue below to make sure that the loop and buffer events
2034 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002035 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002036 }
2037
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002038 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002039 bool active = mState == STATE_ACTIVE;
2040
2041 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2042 bool newUnderrun = false;
2043 if (flags & CBLK_UNDERRUN) {
2044#if 0
2045 // Currently in shared buffer mode, when the server reaches the end of buffer,
2046 // the track stays active in continuous underrun state. It's up to the application
2047 // to pause or stop the track, or set the position to a new offset within buffer.
2048 // This was some experimental code to auto-pause on underrun. Keeping it here
2049 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2050 if (mTransfer == TRANSFER_SHARED) {
2051 mState = STATE_PAUSED;
2052 active = false;
2053 }
2054#endif
2055 if (!mInUnderrun) {
2056 mInUnderrun = true;
2057 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002058 }
2059 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002060
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002061 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002062 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002063
2064 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002065 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002066 Modulo<uint32_t> markerPosition(mMarkerPosition);
2067 // uses 32 bit wraparound for comparison with position.
2068 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002069 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002070 }
2071
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002072 // Determine number of new position callback(s) that will be needed, while locked
2073 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002074 Modulo<uint32_t> newPosition(mNewPosition);
2075 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002076 // FIXME fails for wraparound, need 64 bits
2077 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002078 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002079 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002080 }
2081
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002082 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002083 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002084 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002085 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002086 if (mRefreshRemaining) {
2087 mRefreshRemaining = false;
2088 mRemainingFrames = notificationFrames;
2089 mRetryOnPartialBuffer = false;
2090 }
2091 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002092 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002093 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002094
Andy Hung53c3b5f2014-12-15 16:42:05 -08002095 // Determine the number of new loop callback(s) that will be needed, while locked.
2096 int loopCountNotifications = 0;
2097 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2098
2099 if (mLoopCount > 0) {
2100 int loopCount;
2101 size_t bufferPosition;
2102 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2103 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2104 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2105 mLoopCountNotified = loopCount; // discard any excess notifications
2106 } else if (mLoopCount < 0) {
2107 // FIXME: We're not accurate with notification count and position with infinite looping
2108 // since loopCount from server side will always return -1 (we could decrement it).
2109 size_t bufferPosition = mStaticProxy->getBufferPosition();
2110 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2111 loopPeriod = mLoopEnd - bufferPosition;
2112 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2113 size_t bufferPosition = mStaticProxy->getBufferPosition();
2114 loopPeriod = mFrameCount - bufferPosition;
2115 }
2116
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002117 // These fields don't need to be cached, because they are assigned only by set():
Andy Hungabdb9902015-01-12 15:08:22 -08002118 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002119 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2120
2121 mLock.unlock();
2122
Andy Hunga7f03352015-05-31 21:54:49 -07002123 // get anchor time to account for callbacks.
2124 const nsecs_t timeBeforeCallbacks = systemTime();
2125
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002126 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002127 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2128 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2129 // (and make sure we don't callback for more data while we're stopping).
2130 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002131 struct timespec timeout;
2132 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2133 timeout.tv_nsec = 0;
2134
Glenn Kasten96f04882013-09-20 09:28:56 -07002135 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002136 switch (status) {
2137 case NO_ERROR:
2138 case DEAD_OBJECT:
2139 case TIMED_OUT:
Andy Hung39609a02015-09-03 16:38:38 -07002140 if (status != DEAD_OBJECT) {
2141 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2142 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
2143 mCbf(EVENT_STREAM_END, mUserData, NULL);
2144 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002145 {
2146 AutoMutex lock(mLock);
2147 // The previously assigned value of waitStreamEnd is no longer valid,
2148 // since the mutex has been unlocked and either the callback handler
2149 // or another thread could have re-started the AudioTrack during that time.
2150 waitStreamEnd = mState == STATE_STOPPING;
2151 if (waitStreamEnd) {
2152 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002153 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002154 }
2155 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002156 if (waitStreamEnd && status != DEAD_OBJECT) {
2157 return NS_INACTIVE;
2158 }
2159 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002160 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002161 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002162 }
2163
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002164 // perform callbacks while unlocked
2165 if (newUnderrun) {
2166 mCbf(EVENT_UNDERRUN, mUserData, NULL);
2167 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002168 while (loopCountNotifications > 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002169 mCbf(EVENT_LOOP_END, mUserData, NULL);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002170 --loopCountNotifications;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002171 }
2172 if (flags & CBLK_BUFFER_END) {
2173 mCbf(EVENT_BUFFER_END, mUserData, NULL);
2174 }
2175 if (markerReached) {
2176 mCbf(EVENT_MARKER, mUserData, &markerPosition);
2177 }
2178 while (newPosCount > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002179 size_t temp = newPosition.value(); // FIXME size_t != uint32_t
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002180 mCbf(EVENT_NEW_POS, mUserData, &temp);
2181 newPosition += updatePeriod;
2182 newPosCount--;
2183 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002184
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002185 if (mObservedSequence != sequence) {
2186 mObservedSequence = sequence;
2187 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002188 // for offloaded tracks, just wait for the upper layers to recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002189 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002190 return NS_INACTIVE;
2191 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002192 }
2193
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002194 // if inactive, then don't run me again until re-started
2195 if (!active) {
2196 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002197 }
2198
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002199 // Compute the estimated time until the next timed event (position, markers, loops)
2200 // FIXME only for non-compressed audio
2201 uint32_t minFrames = ~0;
2202 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002203 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002204 }
2205 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002206 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002207 minFrames = loopPeriod;
2208 }
Andy Hung2d85f092015-01-07 12:45:13 -08002209 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002210 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002211 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002212
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002213 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2214 static const uint32_t kPoll = 0;
2215 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2216 minFrames = kPoll * notificationFrames;
2217 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002218
Andy Hunga7f03352015-05-31 21:54:49 -07002219 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2220 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2221 const nsecs_t timeAfterCallbacks = systemTime();
2222
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002223 // Convert frame units to time units
2224 nsecs_t ns = NS_WHENEVER;
2225 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002226 // AudioFlinger consumption of client data may be irregular when coming out of device
2227 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2228 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2229 // half (but no more than half a second) to improve callback accuracy during these temporary
2230 // data surges.
2231 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2232 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2233 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002234 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2235 // TODO: Should we warn if the callback time is too long?
2236 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002237 }
2238
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002239 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2240 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002241 return ns;
2242 }
2243
Andy Hunga7f03352015-05-31 21:54:49 -07002244 // EVENT_MORE_DATA callback handling.
2245 // Timing for linear pcm audio data formats can be derived directly from the
2246 // buffer fill level.
2247 // Timing for compressed data is not directly available from the buffer fill level,
2248 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2249 // to return a certain fill level.
2250
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002251 struct timespec timeout;
2252 const struct timespec *requested = &ClientProxy::kForever;
2253 if (ns != NS_WHENEVER) {
2254 timeout.tv_sec = ns / 1000000000LL;
2255 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002256 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002257 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002258 requested = &timeout;
2259 }
2260
Andy Hungea2b9c02016-02-12 17:06:53 -08002261 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002262 while (mRemainingFrames > 0) {
2263
2264 Buffer audioBuffer;
2265 audioBuffer.frameCount = mRemainingFrames;
2266 size_t nonContig;
2267 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2268 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002269 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002270 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002271 requested = &ClientProxy::kNonBlocking;
2272 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002273 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002274 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002275 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002276 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
2277 (isOffloaded() && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002278 // FIXME bug 25195759
2279 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002280 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002281 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002282 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002283 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002284 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002285
Phil Burkfdb3c072016-02-09 10:47:02 -08002286 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002287 mRetryOnPartialBuffer = false;
2288 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002289 if (ns > 0) { // account for obtain time
2290 const nsecs_t timeNow = systemTime();
2291 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2292 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002293
2294 // delayNs is first computed by the additional frames required in the buffer.
2295 nsecs_t delayNs = framesToNanoseconds(
2296 mRemainingFrames - avail, sampleRate, speed);
2297
2298 // afNs is the AudioFlinger mixer period in ns.
2299 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2300
2301 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2302 // we may have a race if we wait based on the number of frames desired.
2303 // This is a possible issue with resampling and AAudio.
2304 //
2305 // The granularity of audioflinger processing is one mixer period; if
2306 // our wait time is less than one mixer period, wait at most half the period.
2307 if (delayNs < afNs) {
2308 delayNs = std::min(delayNs, afNs / 2);
2309 }
2310
2311 // adjust our ns wait by delayNs.
2312 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2313 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002314 }
2315 return ns;
2316 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002317 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002318
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002319 size_t reqSize = audioBuffer.size;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002320 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2321 // when notifying client it can write more data, pass the total size that can be
2322 // written in the next write() call, since it's not passed through the callback
2323 audioBuffer.size += nonContig;
2324 }
2325 mCbf(mTransfer == TRANSFER_CALLBACK ? EVENT_MORE_DATA : EVENT_CAN_WRITE_MORE_DATA,
2326 mUserData, &audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002327 size_t writtenSize = audioBuffer.size;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002328
Jiabin Huang447cea72020-07-28 22:35:18 +00002329 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002330 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002331 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002332 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002333 return NS_NEVER;
2334 }
2335
2336 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002337 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2338 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2339 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2340 // it only signals to the Java client that it can provide more data, which
2341 // this track is read to accept now.
2342 // The playback thread will be awaken at the next ::write()
2343 return NS_WHENEVER;
2344 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002345 // The callback is done filling buffers
2346 // Keep this thread going to handle timed events and
2347 // still try to get more data in intervals of WAIT_PERIOD_MS
2348 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002349
2350 // mCbf(EVENT_MORE_DATA, ...) might either
2351 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2352 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2353 // (3) Return 0 size when no data is available, does not wait for more data.
2354 //
2355 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2356 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2357 // especially for case (3).
2358 //
2359 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2360 // and this loop; whereas for case (3) we could simply check once with the full
2361 // buffer size and skip the loop entirely.
2362
2363 nsecs_t myns;
Phil Burkfdb3c072016-02-09 10:47:02 -08002364 if (audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002365 // time to wait based on buffer occupancy
2366 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2367 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2368 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002369 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002370 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2371 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2372 myns = datans + (afns / 2);
2373 } else {
2374 // FIXME: This could ping quite a bit if the buffer isn't full.
2375 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2376 myns = kWaitPeriodNs;
2377 }
2378 if (ns > 0) { // account for obtain and callback time
2379 const nsecs_t timeNow = systemTime();
2380 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2381 }
2382 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2383 ns = myns;
2384 }
2385 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002386 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002387
Glenn Kasten138d6f92015-03-20 10:54:51 -07002388 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002389 audioBuffer.frameCount = releasedFrames;
2390 mRemainingFrames -= releasedFrames;
2391 if (misalignment >= releasedFrames) {
2392 misalignment -= releasedFrames;
2393 } else {
2394 misalignment = 0;
2395 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002396
2397 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002398 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002399
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002400 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2401 // if callback doesn't like to accept the full chunk
2402 if (writtenSize < reqSize) {
2403 continue;
2404 }
2405
2406 // There could be enough non-contiguous frames available to satisfy the remaining request
2407 if (mRemainingFrames <= nonContig) {
2408 continue;
2409 }
2410
2411#if 0
2412 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2413 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2414 // that total to a sum == notificationFrames.
2415 if (0 < misalignment && misalignment <= mRemainingFrames) {
2416 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002417 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002418 }
2419#endif
2420
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002421 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002422 if (writtenFrames > 0) {
2423 AutoMutex lock(mLock);
2424 mFramesWritten += writtenFrames;
2425 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002426 mRemainingFrames = notificationFrames;
2427 mRetryOnPartialBuffer = true;
2428
2429 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2430 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002431}
2432
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002433status_t AudioTrack::restoreTrack_l(const char *from)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002434{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002435 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2436 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002437 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002438 mediametrics::LogItem(mMetricsId)
2439 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002440 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002441 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2442 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2443 .set(AMEDIAMETRICS_PROP_WHERE, from)
2444 .record(); });
2445
Andy Hungfb8ede22018-09-12 19:03:24 -07002446 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002447 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002448 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002449
Glenn Kastena47f3162012-11-07 10:13:08 -08002450 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002451 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002452 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002453
Ronghua Wufaeb0f22015-05-21 12:20:21 -07002454 if (isOffloadedOrDirect_l() || mDoNotReconnect) {
Andy Hung1f1db832015-06-08 13:26:10 -07002455 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
2456 // reconsider enabling for linear PCM encodings when position can be preserved.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002457 result = DEAD_OBJECT;
2458 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002459 }
2460
Phil Burk2812d9e2016-01-04 10:34:30 -08002461 // Save so we can return count since creation.
2462 mUnderrunCountOffset = getUnderrunCount_l();
2463
Glenn Kasten200092b2014-08-15 15:13:30 -07002464 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002465 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002466 size_t bufferPosition = 0;
2467 int loopCount = 0;
2468 if (mStaticProxy != 0) {
2469 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002470 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002471 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002472
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002473 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2474 // causes a lot of churn on the service side, and it can reject starting
2475 // playback of a previously created track. May also apply to other cases.
2476 const int INITIAL_RETRIES = 3;
2477 int retries = INITIAL_RETRIES;
2478retry:
2479 if (retries < INITIAL_RETRIES) {
2480 // See the comment for clearAudioConfigCache at the start of the function.
2481 AudioSystem::clearAudioConfigCache();
2482 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002483 mFlags = mOrigFlags;
2484
Glenn Kasten200092b2014-08-15 15:13:30 -07002485 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002486 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002487 // It will also delete the strong references on previous IAudioTrack and IMemory.
2488 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002489 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002490
Eric Laurent6ec546d2018-10-10 16:52:14 -07002491 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002492 // take the frames that will be lost by track recreation into account in saved position
2493 // For streaming tracks, this is the amount we obtained from the user/client
2494 // (not the number actually consumed at the server - those are already lost).
2495 if (mStaticProxy == 0) {
2496 mPosition = mReleased;
2497 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002498 // Continue playback from last known position and restore loop.
2499 if (mStaticProxy != 0) {
2500 if (loopCount != 0) {
2501 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2502 mLoopStart, mLoopEnd, loopCount);
2503 } else {
2504 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002505 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002506 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002507 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002508 }
2509 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002510 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002511 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2512 sp<VolumeShaper::Operation> operationToEnd =
2513 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002514 // TODO: Ideally we would restore to the exact xOffset position
2515 // as returned by getVolumeShaperState(), but we don't have that
2516 // information when restoring at the client unless we periodically poll
2517 // the server or create shared memory state.
2518 //
Andy Hung39399b62017-04-21 15:07:45 -07002519 // For now, we simply advance to the end of the VolumeShaper effect
2520 // if it has been started.
2521 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002522 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002523 }
2524 return mAudioTrack->applyVolumeShaper(shaper.mConfiguration, operationToEnd);
Andy Hung4ef88d72017-02-21 19:47:53 -08002525 });
2526
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002527 if (mState == STATE_ACTIVE) {
Glenn Kastena47f3162012-11-07 10:13:08 -08002528 result = mAudioTrack->start();
Eric Laurent1703cdf2011-03-07 14:52:59 -08002529 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002530 // server resets to zero so we offset
2531 mFramesWrittenServerOffset =
2532 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2533 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002534 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002535 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002536 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002537 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002538 // leave time for an eventual race condition to clear before retrying
2539 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002540 goto retry;
2541 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002542 // if no retries left, set invalid bit to force restoring at next occasion
2543 // and avoid inconsistent active state on client and server sides
2544 if (mCblk != nullptr) {
2545 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2546 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002547 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002548 return result;
2549}
2550
Andy Hung90e8a972015-11-09 16:42:40 -08002551Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002552{
2553 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002554 Modulo<uint32_t> newServer(mProxy->getPosition());
2555 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07002556 // TODO There is controversy about whether there can be "negative jitter" in server position.
2557 // This should be investigated further, and if possible, it should be addressed.
2558 // A more definite failure mode is infrequent polling by client.
2559 // One could call (void)getPosition_l() in releaseBuffer(),
2560 // so mReleased and mPosition are always lock-step as best possible.
2561 // That should ensure delta never goes negative for infrequent polling
2562 // unless the server has more than 2^31 frames in its buffer,
2563 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08002564 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07002565 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08002566 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07002567 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08002568 if (delta > 0) { // avoid retrograde
2569 mPosition += delta;
2570 }
2571 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07002572}
2573
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002574bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07002575{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002576 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07002577 // applicable for mixing tracks only (not offloaded or direct)
2578 if (mStaticProxy != 0) {
2579 return true; // static tracks do not have issues with buffer sizing.
2580 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07002581 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08002582 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
2583 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002584 const bool allowed = mFrameCount >= minFrameCount;
2585 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07002586 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002587 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
2588 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002589 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002590 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07002591 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002592 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002593}
2594
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00002595status_t AudioTrack::setParameters(const String8& keyValuePairs)
2596{
2597 AutoMutex lock(mLock);
Glenn Kasten53cec222013-08-29 09:01:02 -07002598 return mAudioTrack->setParameters(keyValuePairs);
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00002599}
2600
Dean Wheatleya70eef72018-01-04 14:23:50 +11002601status_t AudioTrack::selectPresentation(int presentationId, int programId)
2602{
2603 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08002604 AudioParameter param = AudioParameter();
2605 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
2606 param.addInt(String8(AudioParameter::keyProgramId), programId);
2607 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
2608 __func__, mPortId, param.toString().string());
2609
2610 return mAudioTrack->setParameters(param.toString());
Dean Wheatleya70eef72018-01-04 14:23:50 +11002611}
2612
Andy Hung9fc8b5c2017-01-24 13:36:48 -08002613VolumeShaper::Status AudioTrack::applyVolumeShaper(
2614 const sp<VolumeShaper::Configuration>& configuration,
2615 const sp<VolumeShaper::Operation>& operation)
2616{
2617 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08002618 mVolumeHandler->setIdIfNecessary(configuration);
Andy Hung9fc8b5c2017-01-24 13:36:48 -08002619 VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07002620
2621 if (status == DEAD_OBJECT) {
2622 if (restoreTrack_l("applyVolumeShaper") == OK) {
2623 status = mAudioTrack->applyVolumeShaper(configuration, operation);
2624 }
2625 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002626 if (status >= 0) {
2627 // save VolumeShaper for restore
2628 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07002629 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
2630 mVolumeHandler->setStarted();
2631 }
2632 } else {
2633 // warn only if not an expected restore failure.
2634 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08002635 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08002636 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08002637 return status;
2638}
2639
2640sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
2641{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08002642 AutoMutex lock(mLock);
Andy Hung39399b62017-04-21 15:07:45 -07002643 sp<VolumeShaper::State> state = mAudioTrack->getVolumeShaperState(id);
2644 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
2645 if (restoreTrack_l("getVolumeShaperState") == OK) {
2646 state = mAudioTrack->getVolumeShaperState(id);
2647 }
2648 }
2649 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08002650}
2651
Andy Hungea2b9c02016-02-12 17:06:53 -08002652status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
2653{
2654 if (timestamp == nullptr) {
2655 return BAD_VALUE;
2656 }
2657 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07002658 return getTimestamp_l(timestamp);
2659}
2660
2661status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
2662{
Andy Hungea2b9c02016-02-12 17:06:53 -08002663 if (mCblk->mFlags & CBLK_INVALID) {
2664 const status_t status = restoreTrack_l("getTimestampExtended");
2665 if (status != OK) {
2666 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
2667 // recommending that the track be recreated.
2668 return DEAD_OBJECT;
2669 }
2670 }
2671 // check for offloaded/direct here in case restoring somehow changed those flags.
2672 if (isOffloadedOrDirect_l()) {
2673 return INVALID_OPERATION; // not supported
2674 }
2675 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07002676 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08002677 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08002678 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07002679 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
2680 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
2681 // server side frame offset in case AudioTrack has been restored.
2682 for (int i = ExtendedTimestamp::LOCATION_SERVER;
2683 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
2684 if (timestamp->mTimeNs[i] >= 0) {
2685 // apply server offset (frames flushed is ignored
2686 // so we don't report the jump when the flush occurs).
2687 timestamp->mPosition[i] += mFramesWrittenServerOffset;
2688 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08002689 }
2690 }
2691 return found ? OK : WOULD_BLOCK;
2692}
2693
Glenn Kastence703742013-07-19 16:33:58 -07002694status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
2695{
Glenn Kasten53cec222013-08-29 09:01:02 -07002696 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07002697 return getTimestamp_l(timestamp);
2698}
Phil Burk1b420972015-04-22 10:52:21 -07002699
Andy Hung65ffdfc2016-10-10 15:52:11 -07002700status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
2701{
Phil Burk1b420972015-04-22 10:52:21 -07002702 bool previousTimestampValid = mPreviousTimestampValid;
2703 // Set false here to cover all the error return cases.
2704 mPreviousTimestampValid = false;
2705
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002706 switch (mState) {
2707 case STATE_ACTIVE:
2708 case STATE_PAUSED:
2709 break; // handle below
2710 case STATE_FLUSHED:
2711 case STATE_STOPPED:
2712 return WOULD_BLOCK;
2713 case STATE_STOPPING:
2714 case STATE_PAUSED_STOPPING:
2715 if (!isOffloaded_l()) {
2716 return INVALID_OPERATION;
2717 }
2718 break; // offloaded tracks handled below
2719 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07002720 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08002721 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002722 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07002723 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002724
Eric Laurent275e8e92014-11-30 15:14:47 -08002725 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07002726 const status_t status = restoreTrack_l("getTimestamp");
2727 if (status != OK) {
2728 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
2729 // recommending that the track be recreated.
2730 return DEAD_OBJECT;
2731 }
Eric Laurent275e8e92014-11-30 15:14:47 -08002732 }
2733
Glenn Kasten200092b2014-08-15 15:13:30 -07002734 // The presented frame count must always lag behind the consumed frame count.
2735 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08002736
2737 status_t status;
Andy Hung818e7a32016-02-16 18:08:07 -08002738 if (isOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08002739 // use Binder to get timestamp
2740 status = mAudioTrack->getTimestamp(timestamp);
2741 } else {
2742 // read timestamp from shared memory
2743 ExtendedTimestamp ets;
2744 status = mProxy->getTimestamp(&ets);
2745 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07002746 ExtendedTimestamp::Location location;
2747 status = ets.getBestTimestamp(&timestamp, &location);
2748
2749 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002750 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07002751 // It is possible that the best location has moved from the kernel to the server.
2752 // In this case we adjust the position from the previous computed latency.
2753 if (location == ExtendedTimestamp::LOCATION_SERVER) {
2754 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07002755 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08002756 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07002757 // check that the last kernel OK time info exists and the positions
2758 // are valid (if they predate the current track, the positions may
2759 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07002760 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07002761 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07002762 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
2763 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
2764 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07002765 ?
2766 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
2767 / 1000)
2768 :
2769 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
2770 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07002771 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08002772 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07002773 if (frames >= ets.mPosition[location]) {
2774 timestamp.mPosition = 0;
2775 } else {
2776 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
2777 }
Andy Hung69488c42016-05-16 18:43:33 -07002778 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
2779 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07002780 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08002781 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07002782
2783 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
2784 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
2785 // In Q, we don't return errors as an invalid time
2786 // but instead we leave the last kernel good timestamp alone.
2787 //
2788 // If server is identical to kernel, the device data pipeline is idle.
2789 // A better start time is now. The retrograde check ensures
2790 // timestamp monotonicity.
2791 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07002792 if (!mTimestampStallReported) {
2793 ALOGD("%s(%d): device stall time corrected using current time %lld",
2794 __func__, mPortId, (long long)nowNs);
2795 mTimestampStallReported = true;
2796 }
Andy Hung98731a22019-04-08 19:19:07 -07002797 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07002798 } else {
2799 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07002800 }
Andy Hungb01faa32016-04-27 12:51:32 -07002801 }
Andy Hung5d313802016-10-10 15:09:39 -07002802
2803 // We update the timestamp time even when paused.
2804 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
2805 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07002806 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07002807 const int64_t lag =
2808 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
2809 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
2810 ? int64_t(mAfLatency * 1000000LL)
2811 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
2812 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
2813 * NANOS_PER_SECOND / mSampleRate;
2814 const int64_t limit = now - lag; // no earlier than this limit
2815 if (at < limit) {
2816 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
2817 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07002818 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07002819 }
2820 }
Andy Hungb01faa32016-04-27 12:51:32 -07002821 mPreviousLocation = location;
2822 } else {
2823 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08002824 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07002825 }
Andy Hung6ae58432016-02-16 18:32:24 -08002826 }
2827 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07002828 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
2829 // other failures are signaled by a negative time.
2830 // If we come out of FLUSHED or STOPPED where the position is known
2831 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
2832 // "zero" for NuPlayer). We don't convert for track restoration as position
2833 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07002834 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08002835 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07002836 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
2837 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
2838 status = WOULD_BLOCK;
2839 }
Andy Hung6ae58432016-02-16 18:32:24 -08002840 }
2841 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002842 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002843 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002844 return status;
2845 }
2846 if (isOffloadedOrDirect_l()) {
2847 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
2848 // use cached paused position in case another offloaded track is running.
2849 timestamp.mPosition = mPausedPosition;
2850 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07002851 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002852 return NO_ERROR;
2853 }
2854
2855 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07002856 // be asynchronous or return near finish or exhibit glitchy behavior.
2857 //
2858 // Originally this showed up as the first timestamp being a continuation of
2859 // the previous song under gapless playback.
2860 // However, we sometimes see zero timestamps, then a glitch of
2861 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07002862 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002863 static const int kTimeJitterUs = 100000; // 100 ms
2864 static const int k1SecUs = 1000000;
2865
2866 const int64_t timeNow = getNowUs();
2867
Andy Hungffa36952017-08-17 10:41:51 -07002868 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002869 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07002870 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002871 return WOULD_BLOCK; // stale timestamp time, occurs before start.
2872 }
Andy Hungffa36952017-08-17 10:41:51 -07002873 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002874 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002875 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002876
2877 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
2878 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07002879 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002880 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07002881 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07002882 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002883 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08002884 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002885 (long long)deltaTimeUs, (long long)deltaPositionByUs,
2886 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07002887 mTimestampStartupGlitchReported = true;
2888 if (previousTimestampValid
2889 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
2890 timestamp = mPreviousTimestamp;
2891 mPreviousTimestampValid = true;
2892 return NO_ERROR;
2893 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002894 return WOULD_BLOCK;
2895 }
Andy Hungc8e09c62015-06-03 23:43:36 -07002896 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07002897 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07002898 }
2899 } else {
Andy Hungffa36952017-08-17 10:41:51 -07002900 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002901 }
Andy Hungc8e09c62015-06-03 23:43:36 -07002902 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07002903 }
2904 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07002905 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
2906 (void) updateAndGetPosition_l();
2907 // Server consumed (mServer) and presented both use the same server time base,
2908 // and server consumed is always >= presented.
2909 // The delta between these represents the number of frames in the buffer pipeline.
2910 // If this delta between these is greater than the client position, it means that
2911 // actually presented is still stuck at the starting line (figuratively speaking),
2912 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08002913 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
2914 // mPosition exceeds 32 bits.
2915 // TODO Remove when timestamp is updated to contain pipeline status info.
2916 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
2917 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
2918 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07002919 return INVALID_OPERATION;
2920 }
2921 // Convert timestamp position from server time base to client time base.
2922 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
2923 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08002924 // Use Modulo computation here.
2925 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07002926 // Immediately after a call to getPosition_l(), mPosition and
2927 // mServer both represent the same frame position. mPosition is
2928 // in client's point of view, and mServer is in server's point of
2929 // view. So the difference between them is the "fudge factor"
2930 // between client and server views due to stop() and/or new
2931 // IAudioTrack. And timestamp.mPosition is initially in server's
2932 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07002933 }
Phil Burk1b420972015-04-22 10:52:21 -07002934
2935 // Prevent retrograde motion in timestamp.
2936 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
2937 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07002938 // Fix stale time when checking timestamp right after start().
2939 // The position is at the last reported location but the time can be stale
2940 // due to pause or standby or cold start latency.
2941 //
2942 // We keep advancing the time (but not the position) to ensure that the
2943 // stale value does not confuse the application.
2944 //
2945 // For offload compatibility, use a default lag value here.
2946 // Any time discrepancy between this update and the pause timestamp is handled
2947 // by the retrograde check afterwards.
2948 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
2949 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
2950 const int64_t limitNs = mStartNs - lagNs;
2951 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07002952 if (!mTimestampStaleTimeReported) {
2953 ALOGD("%s(%d): stale timestamp time corrected, "
2954 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
2955 __func__, mPortId,
2956 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
2957 mTimestampStaleTimeReported = true;
2958 }
Andy Hung3b8c6332019-04-03 19:29:36 -07002959 timestamp.mTime = convertNsToTimespec(limitNs);
2960 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07002961 } else {
2962 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07002963 }
2964
Andy Hungffa36952017-08-17 10:41:51 -07002965 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07002966 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07002967 const int64_t previousTimeNanos =
2968 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07002969
2970 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07002971 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07002972 if (!mTimestampRetrogradeTimeReported) {
2973 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
2974 __func__, mPortId,
2975 (long long)currentTimeNanos, (long long)previousTimeNanos);
2976 mTimestampRetrogradeTimeReported = true;
2977 }
Andy Hung5d313802016-10-10 15:09:39 -07002978 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07002979 } else {
2980 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07002981 }
2982
2983 // Looking at signed delta will work even when the timestamps
2984 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08002985 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
2986 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07002987 if (deltaPosition < 0) {
2988 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07002989 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002990 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08002991 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07002992 deltaPosition,
2993 timestamp.mPosition,
2994 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07002995 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07002996 }
2997 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07002998 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07002999 }
Andy Hung5d313802016-10-10 15:09:39 -07003000 if (deltaPosition < 0) {
3001 timestamp.mPosition = mPreviousTimestamp.mPosition;
3002 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003003 }
Andy Hung5d313802016-10-10 15:09:39 -07003004#if 0
3005 // Uncomment this to verify audio timestamp rate.
3006 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003007 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003008 if (deltaTime != 0) {
3009 const int64_t computedSampleRate =
3010 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003011 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003012 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003013 (unsigned)computedSampleRate, mSampleRate);
3014 }
3015#endif
Phil Burk1b420972015-04-22 10:52:21 -07003016 }
3017 mPreviousTimestamp = timestamp;
3018 mPreviousTimestampValid = true;
3019 }
3020
Glenn Kastenfe346c72013-08-30 13:28:22 -07003021 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003022}
3023
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003024String8 AudioTrack::getParameters(const String8& keys)
3025{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003026 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003027 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003028 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003029 } else {
3030 return String8::empty();
3031 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003032}
3033
Glenn Kasten23a75452014-01-13 10:37:17 -08003034bool AudioTrack::isOffloaded() const
3035{
3036 AutoMutex lock(mLock);
3037 return isOffloaded_l();
3038}
3039
Eric Laurentab5cdba2014-06-09 17:22:27 -07003040bool AudioTrack::isDirect() const
3041{
3042 AutoMutex lock(mLock);
3043 return isDirect_l();
3044}
3045
3046bool AudioTrack::isOffloadedOrDirect() const
3047{
3048 AutoMutex lock(mLock);
3049 return isOffloadedOrDirect_l();
3050}
3051
3052
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003053status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003054{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003055 String8 result;
3056
3057 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003058 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003059 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003060 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
3061 (mStreamType == AUDIO_STREAM_DEFAULT) ?
François Gaffie58d4be52018-11-06 15:30:12 +01003062 AudioSystem::attributesToStreamType(mAttributes) :
3063 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003064 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003065 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003066 mFormat, mChannelMask, mChannelCount);
3067 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3068 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3069 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3070 mFrameCount, mReqFrameCount);
3071 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3072 " req. notif. per buff(%u)\n",
3073 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3074 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3075 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3076 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3077 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003078 ::write(fd, result.string(), result.size());
3079 return NO_ERROR;
3080}
3081
Phil Burk2812d9e2016-01-04 10:34:30 -08003082uint32_t AudioTrack::getUnderrunCount() const
3083{
3084 AutoMutex lock(mLock);
3085 return getUnderrunCount_l();
3086}
3087
3088uint32_t AudioTrack::getUnderrunCount_l() const
3089{
3090 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3091}
3092
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003093uint32_t AudioTrack::getUnderrunFrames() const
3094{
3095 AutoMutex lock(mLock);
3096 return mProxy->getUnderrunFrames();
3097}
3098
Eric Laurent296fb132015-05-01 11:38:42 -07003099status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3100{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003101
Eric Laurent296fb132015-05-01 11:38:42 -07003102 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003103 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003104 return BAD_VALUE;
3105 }
3106 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003107 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003108 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003109 return INVALID_OPERATION;
3110 }
3111 status_t status = NO_ERROR;
3112 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3113 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003114 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003115 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003116 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003117 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003118 }
3119 mDeviceCallback = callback;
3120 return status;
3121}
3122
3123status_t AudioTrack::removeAudioDeviceCallback(
3124 const sp<AudioSystem::AudioDeviceCallback>& callback)
3125{
3126 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003127 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003128 return BAD_VALUE;
3129 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003130 AutoMutex lock(mLock);
3131 if (mDeviceCallback.unsafe_get() != callback.get()) {
3132 ALOGW("%s removing different callback!", __FUNCTION__);
3133 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003134 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003135 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003136 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003137 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003138 }
Eric Laurent296fb132015-05-01 11:38:42 -07003139 return NO_ERROR;
3140}
3141
Eric Laurentad2e7b92017-09-14 20:06:42 -07003142
3143void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3144 audio_port_handle_t deviceId)
3145{
3146 sp<AudioSystem::AudioDeviceCallback> callback;
3147 {
3148 AutoMutex lock(mLock);
3149 if (audioIo != mOutput) {
3150 return;
3151 }
3152 callback = mDeviceCallback.promote();
3153 // only update device if the track is active as route changes due to other use cases are
3154 // irrelevant for this client
3155 if (mState == STATE_ACTIVE) {
3156 mRoutedDeviceId = deviceId;
3157 }
3158 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003159
Eric Laurentad2e7b92017-09-14 20:06:42 -07003160 if (callback.get() != nullptr) {
3161 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3162 }
3163}
3164
Andy Hunge13f8a62016-03-30 14:20:42 -07003165status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3166{
3167 if (msec == nullptr ||
3168 (location != ExtendedTimestamp::LOCATION_SERVER
3169 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3170 return BAD_VALUE;
3171 }
3172 AutoMutex lock(mLock);
3173 // inclusive of offloaded and direct tracks.
3174 //
3175 // It is possible, but not enabled, to allow duration computation for non-pcm
3176 // audio_has_proportional_frames() formats because currently they have
3177 // the drain rate equivalent to the pcm sample rate * framesize.
3178 if (!isPurePcmData_l()) {
3179 return INVALID_OPERATION;
3180 }
3181 ExtendedTimestamp ets;
3182 if (getTimestamp_l(&ets) == OK
3183 && ets.mTimeNs[location] > 0) {
3184 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3185 - ets.mPosition[location];
3186 if (diff < 0) {
3187 *msec = 0;
3188 } else {
3189 // ms is the playback time by frames
3190 int64_t ms = (int64_t)((double)diff * 1000 /
3191 ((double)mSampleRate * mPlaybackRate.mSpeed));
3192 // clockdiff is the timestamp age (negative)
3193 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3194 ets.mTimeNs[location]
3195 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3196 - systemTime(SYSTEM_TIME_MONOTONIC);
3197
3198 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3199 static const int NANOS_PER_MILLIS = 1000000;
3200 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3201 }
3202 return NO_ERROR;
3203 }
3204 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3205 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3206 }
3207 // use server position directly (offloaded and direct arrive here)
3208 updateAndGetPosition_l();
3209 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3210 *msec = (diff <= 0) ? 0
3211 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3212 return NO_ERROR;
3213}
3214
Andy Hung65ffdfc2016-10-10 15:52:11 -07003215bool AudioTrack::hasStarted()
3216{
3217 AutoMutex lock(mLock);
3218 switch (mState) {
3219 case STATE_STOPPED:
3220 if (isOffloadedOrDirect_l()) {
3221 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003222 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003223 }
3224 // A normal audio track may still be draining, so
3225 // check if stream has ended. This covers fasttrack position
3226 // instability and start/stop without any data written.
3227 if (mProxy->getStreamEndDone()) {
3228 return true;
3229 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003230 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003231 case STATE_ACTIVE:
3232 case STATE_STOPPING:
3233 break;
3234 case STATE_PAUSED:
3235 case STATE_PAUSED_STOPPING:
3236 case STATE_FLUSHED:
3237 return false; // we're not active
3238 default:
Eric Laurent973db022018-11-20 14:54:31 -08003239 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003240 break;
3241 }
3242
3243 // wait indicates whether we need to wait for a timestamp.
3244 // This is conservatively figured - if we encounter an unexpected error
3245 // then we will not wait.
3246 bool wait = false;
3247 if (isOffloadedOrDirect_l()) {
3248 AudioTimestamp ts;
3249 status_t status = getTimestamp_l(ts);
3250 if (status == WOULD_BLOCK) {
3251 wait = true;
3252 } else if (status == OK) {
3253 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3254 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003255 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003256 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003257 (int)wait,
3258 ts.mPosition,
3259 (long long)mStartTs.mPosition);
3260 } else {
3261 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3262 ExtendedTimestamp ets;
3263 status_t status = getTimestamp_l(&ets);
3264 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3265 wait = true;
3266 } else if (status == OK) {
3267 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3268 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3269 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3270 continue;
3271 }
3272 wait = ets.mPosition[location] == 0
3273 || ets.mPosition[location] == mStartEts.mPosition[location];
3274 break;
3275 }
3276 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003277 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003278 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003279 (int)wait,
3280 (long long)ets.mPosition[location],
3281 (long long)mStartEts.mPosition[location]);
3282 }
3283 return !wait;
3284}
3285
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003286// =========================================================================
3287
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003288void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003289{
3290 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3291 if (audioTrack != 0) {
3292 AutoMutex lock(audioTrack->mLock);
3293 audioTrack->mProxy->binderDied();
3294 }
3295}
3296
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003297// =========================================================================
3298
Andy Hungca353672019-03-06 11:54:38 -08003299AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003300 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3301 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003302 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003303{
3304}
3305
3306AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003307{
3308}
3309
3310bool AudioTrack::AudioTrackThread::threadLoop()
3311{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003312 {
3313 AutoMutex _l(mMyLock);
3314 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003315 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003316 mMyCond.wait(mMyLock);
3317 // caller will check for exitPending()
3318 return true;
3319 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003320 if (mIgnoreNextPausedInt) {
3321 mIgnoreNextPausedInt = false;
3322 mPausedInt = false;
3323 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003324 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003325 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003326 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003327 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003328 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3329 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003330 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003331 mMyCond.wait(mMyLock);
3332 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003333 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003334 return true;
3335 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003336 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003337 if (exitPending()) {
3338 return false;
3339 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003340 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003341 switch (ns) {
3342 case 0:
3343 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003344 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003345 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003346 return true;
3347 case NS_NEVER:
3348 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003349 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003350 // Event driven: call wake() when callback notifications conditions change.
3351 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003352 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003353 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003354 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003355 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003356 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003357 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003358 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003359}
3360
Glenn Kasten3acbd052012-02-28 10:39:56 -08003361void AudioTrack::AudioTrackThread::requestExit()
3362{
3363 // must be in this order to avoid a race condition
3364 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003365 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003366}
3367
3368void AudioTrack::AudioTrackThread::pause()
3369{
3370 AutoMutex _l(mMyLock);
3371 mPaused = true;
3372}
3373
3374void AudioTrack::AudioTrackThread::resume()
3375{
3376 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003377 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003378 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003379 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003380 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003381 mMyCond.signal();
3382 }
3383}
3384
Andy Hung3c09c782014-12-29 18:39:32 -08003385void AudioTrack::AudioTrackThread::wake()
3386{
3387 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003388 if (!mPaused) {
3389 // wake() might be called while servicing a callback - ignore the next
3390 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003391 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003392 if (mPausedInt && mPausedNs > 0) {
3393 // audio track is active and internally paused with timeout.
3394 mPausedInt = false;
3395 mMyCond.signal();
3396 }
Andy Hung3c09c782014-12-29 18:39:32 -08003397 }
3398}
3399
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003400void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3401{
3402 AutoMutex _l(mMyLock);
3403 mPausedInt = true;
3404 mPausedNs = ns;
3405}
3406
jiabinf6eb4c32020-02-25 14:06:25 -08003407binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3408 const std::vector<uint8_t>& audioMetadata)
3409{
3410 AutoMutex _l(mAudioTrackCbLock);
3411 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3412 if (callback.get() != nullptr) {
3413 callback->onCodecFormatChanged(audioMetadata);
3414 } else {
3415 mCallback.clear();
3416 }
3417 return binder::Status::ok();
3418}
3419
3420void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3421 const sp<media::IAudioTrackCallback> &callback) {
3422 AutoMutex lock(mAudioTrackCbLock);
3423 mCallback = callback;
3424}
3425
Glenn Kasten40bc9062015-03-20 09:09:33 -07003426} // namespace android