[RESTRICT AUTOMERGE] CTS test for Android Security b/34749571
Bug: 34749571
Bug: 72459536
Test: Ran the new testcase on android-10.0.0_r1 with/without patch
Change-Id: I137dfce18d847e3287d94c03a24cdb21bf59032e
diff --git a/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/Android.bp b/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/Android.bp
new file mode 100644
index 0000000..aca99a5
--- /dev/null
+++ b/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/Android.bp
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+cc_test {
+ name: "CVE-2017-0597",
+ defaults: ["cts_hostsidetests_securitybulletin_defaults"],
+ srcs: [
+ "poc.cpp",
+ "AudioTrackModified.cpp",
+ ],
+ include_dirs: [
+ "frameworks/av/services/audioflinger",
+ "frameworks/av/media/libaudioclient/include/media",
+ "system/media/audio/include/system",
+ "system/media/audio_utils/include",
+ ],
+ shared_libs: [
+ "libc",
+ "liblog",
+ "libcutils",
+ "libutils",
+ "libbinder",
+ "libaudioutils",
+ "libaudioclient",
+ "libmediautils",
+ "libmedia_helper",
+ "libmediametrics",
+ "libnblog",
+ "libprocessgroup",
+ ],
+ cppflags: [
+ "-Wall",
+ "-Werror",
+ "-Wno-error=deprecated-declarations",
+ "-Wunused",
+ "-Wunreachable-code",
+ ],
+}
diff --git a/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/AudioTrackModified.cpp b/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/AudioTrackModified.cpp
new file mode 100644
index 0000000..4f07713
--- /dev/null
+++ b/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/AudioTrackModified.cpp
@@ -0,0 +1,3252 @@
+/**
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+/* This file is taken from frameworks/av/media/libaudioclient and is modified
+ * for CVE-2017-0597 */
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioTrack"
+
+#include <inttypes.h>
+#include <math.h>
+#include <sys/resource.h>
+
+#include <android-base/macros.h>
+#include <audio_utils/clock.h>
+#include <audio_utils/primitives.h>
+#include <binder/IPCThreadState.h>
+#include <media/AudioTrack.h>
+#include <utils/Log.h>
+#include <private/media/AudioTrackShared.h>
+#include <processgroup/sched_policy.h>
+#include <media/IAudioFlinger.h>
+#include <media/IAudioPolicyService.h>
+#include <media/AudioParameter.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/AudioSystem.h>
+#include <media/MediaAnalyticsItem.h>
+#include <media/TypeConverter.h>
+
+#define WAIT_PERIOD_MS 10
+#define WAIT_STREAM_END_TIMEOUT_SEC 120
+static const int kMaxLoopCountNotifications = 32;
+
+namespace android {
+// ---------------------------------------------------------------------------
+
+using media::VolumeShaper;
+
+// TODO: Move to a separate .h
+
+template <typename T>
+static inline const T &min(const T &x, const T &y) {
+ return x < y ? x : y;
+}
+
+template <typename T>
+static inline const T &max(const T &x, const T &y) {
+ return x > y ? x : y;
+}
+
+static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
+{
+ return ((double)frames * 1000000000) / ((double)sampleRate * speed);
+}
+
+static int64_t convertTimespecToUs(const struct timespec &tv)
+{
+ return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
+}
+
+// TODO move to audio_utils.
+static inline struct timespec convertNsToTimespec(int64_t ns) {
+ struct timespec tv;
+ tv.tv_sec = static_cast<time_t>(ns / NANOS_PER_SECOND);
+ tv.tv_nsec = static_cast<long>(ns % NANOS_PER_SECOND);
+ return tv;
+}
+
+// current monotonic time in microseconds.
+static int64_t getNowUs()
+{
+ struct timespec tv;
+ (void) clock_gettime(CLOCK_MONOTONIC, &tv);
+ return convertTimespecToUs(tv);
+}
+
+// FIXME: we don't use the pitch setting in the time stretcher (not working);
+// instead we emulate it using our sample rate converter.
+static const bool kFixPitch = true; // enable pitch fix
+static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)
+{
+ return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate;
+}
+
+static inline float adjustSpeed(float speed, float pitch)
+{
+ return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
+}
+
+static inline float adjustPitch(float pitch)
+{
+ return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
+}
+
+// static
+status_t AudioTrack::getMinFrameCount(
+ size_t* frameCount,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate)
+{
+ if (frameCount == NULL) {
+ return BAD_VALUE;
+ }
+
+ // FIXME handle in server, like createTrack_l(), possible missing info:
+ // audio_io_handle_t output
+ // audio_format_t format
+ // audio_channel_mask_t channelMask
+ // audio_output_flags_t flags (FAST)
+ uint32_t afSampleRate;
+ status_t status;
+ status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
+ if (status != NO_ERROR) {
+ ALOGE("%s(): Unable to query output sample rate for stream type %d; status %d",
+ __func__, streamType, status);
+ return status;
+ }
+ size_t afFrameCount;
+ status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
+ if (status != NO_ERROR) {
+ ALOGE("%s(): Unable to query output frame count for stream type %d; status %d",
+ __func__, streamType, status);
+ return status;
+ }
+ uint32_t afLatency;
+ status = AudioSystem::getOutputLatency(&afLatency, streamType);
+ if (status != NO_ERROR) {
+ ALOGE("%s(): Unable to query output latency for stream type %d; status %d",
+ __func__, streamType, status);
+ return status;
+ }
+
+ // When called from createTrack, speed is 1.0f (normal speed).
+ // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
+ *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
+ sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
+
+ // The formula above should always produce a non-zero value under normal circumstances:
+ // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
+ // Return error in the unlikely event that it does not, as that's part of the API contract.
+ if (*frameCount == 0) {
+ ALOGE("%s(): failed for streamType %d, sampleRate %u",
+ __func__, streamType, sampleRate);
+ return BAD_VALUE;
+ }
+ ALOGV("%s(): getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
+ __func__, *frameCount, afFrameCount, afSampleRate, afLatency);
+ return NO_ERROR;
+}
+
+// static
+bool AudioTrack::isDirectOutputSupported(const audio_config_base_t& config,
+ const audio_attributes_t& attributes) {
+ ALOGV("%s()", __FUNCTION__);
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return false;
+ return aps->isDirectOutputSupported(config, attributes);
+}
+
+// ---------------------------------------------------------------------------
+
+void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
+{
+ // only if we're in a good state...
+ // XXX: shall we gather alternative info if failing?
+ const status_t lstatus = track->initCheck();
+ if (lstatus != NO_ERROR) {
+ ALOGD("%s(): no metrics gathered, track status=%d", __func__, (int) lstatus);
+ return;
+ }
+
+#define MM_PREFIX "android.media.audiotrack." // avoid cut-n-paste errors.
+
+ // Java API 28 entries, do not change.
+ mAnalyticsItem->setCString(MM_PREFIX "streamtype", toString(track->streamType()).c_str());
+ mAnalyticsItem->setCString(MM_PREFIX "type",
+ toString(track->mAttributes.content_type).c_str());
+ mAnalyticsItem->setCString(MM_PREFIX "usage", toString(track->mAttributes.usage).c_str());
+
+ // Non-API entries, these can change due to a Java string mistake.
+ mAnalyticsItem->setInt32(MM_PREFIX "sampleRate", (int32_t)track->mSampleRate);
+ mAnalyticsItem->setInt64(MM_PREFIX "channelMask", (int64_t)track->mChannelMask);
+ // Non-API entries, these can change.
+ mAnalyticsItem->setInt32(MM_PREFIX "portId", (int32_t)track->mPortId);
+ mAnalyticsItem->setCString(MM_PREFIX "encoding", toString(track->mFormat).c_str());
+ mAnalyticsItem->setInt32(MM_PREFIX "frameCount", (int32_t)track->mFrameCount);
+ mAnalyticsItem->setCString(MM_PREFIX "attributes", toString(track->mAttributes).c_str());
+}
+
+// hand the user a snapshot of the metrics.
+status_t AudioTrack::getMetrics(MediaAnalyticsItem * &item)
+{
+ mMediaMetrics.gather(this);
+ MediaAnalyticsItem *tmp = mMediaMetrics.dup();
+ if (tmp == nullptr) {
+ return BAD_VALUE;
+ }
+ item = tmp;
+ return NO_ERROR;
+}
+
+AudioTrack::AudioTrack()
+ : mStatus(NO_INIT),
+ mState(STATE_STOPPED),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(SP_DEFAULT),
+ mPausedPosition(0),
+ mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
+ mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE)
+{
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
+ mAttributes.usage = AUDIO_USAGE_UNKNOWN;
+ mAttributes.flags = 0x0;
+ strcpy(mAttributes.tags, "");
+}
+
+AudioTrack::AudioTrack(
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ audio_output_flags_t flags,
+ callback_t cbf,
+ void* user,
+ int32_t notificationFrames,
+ audio_session_t sessionId,
+ transfer_type transferType,
+ const audio_offload_info_t *offloadInfo,
+ uid_t uid,
+ pid_t pid,
+ const audio_attributes_t* pAttributes,
+ bool doNotReconnect,
+ float maxRequiredSpeed,
+ audio_port_handle_t selectedDeviceId)
+ : mStatus(NO_INIT),
+ mState(STATE_STOPPED),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(SP_DEFAULT),
+ mPausedPosition(0)
+{
+ mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
+
+ (void)set(streamType, sampleRate, format, channelMask,
+ frameCount, flags, cbf, user, notificationFrames,
+ 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
+ offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
+}
+
+AudioTrack::AudioTrack(
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ const sp<IMemory>& sharedBuffer,
+ audio_output_flags_t flags,
+ callback_t cbf,
+ void* user,
+ int32_t notificationFrames,
+ audio_session_t sessionId,
+ transfer_type transferType,
+ const audio_offload_info_t *offloadInfo,
+ uid_t uid,
+ pid_t pid,
+ const audio_attributes_t* pAttributes,
+ bool doNotReconnect,
+ float maxRequiredSpeed)
+ : mStatus(NO_INIT),
+ mState(STATE_STOPPED),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(SP_DEFAULT),
+ mPausedPosition(0),
+ mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
+{
+ mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
+
+ (void)set(streamType, sampleRate, format, channelMask,
+ // Modification for CVE-2017-0597 START
+ 0xFFFFFF00 /*frameCount*/,
+ // Modification for CVE-2017-0597 END
+ flags, cbf, user, notificationFrames,
+ sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
+ uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
+}
+
+AudioTrack::~AudioTrack()
+{
+ // pull together the numbers, before we clean up our structures
+ mMediaMetrics.gather(this);
+
+ if (mStatus == NO_ERROR) {
+ // Make sure that callback function exits in the case where
+ // it is looping on buffer full condition in obtainBuffer().
+ // Otherwise the callback thread will never exit.
+ stop();
+ if (mAudioTrackThread != 0) {
+ mProxy->interrupt();
+ mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
+ mAudioTrackThread->requestExitAndWait();
+ mAudioTrackThread.clear();
+ }
+ // No lock here: worst case we remove a NULL callback which will be a nop
+ if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
+ AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
+ }
+ IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
+ mAudioTrack.clear();
+ mCblkMemory.clear();
+ mSharedBuffer.clear();
+ IPCThreadState::self()->flushCommands();
+ ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
+ __func__, mPortId,
+ mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid);
+ AudioSystem::releaseAudioSessionId(mSessionId, mClientPid);
+ }
+}
+
+status_t AudioTrack::set(
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ audio_output_flags_t flags,
+ callback_t cbf,
+ void* user,
+ int32_t notificationFrames,
+ const sp<IMemory>& sharedBuffer,
+ bool threadCanCallJava,
+ audio_session_t sessionId,
+ transfer_type transferType,
+ const audio_offload_info_t *offloadInfo,
+ uid_t uid,
+ pid_t pid,
+ const audio_attributes_t* pAttributes,
+ bool doNotReconnect,
+ float maxRequiredSpeed,
+ audio_port_handle_t selectedDeviceId)
+{
+ status_t status;
+ uint32_t channelCount;
+ pid_t callingPid;
+ pid_t myPid;
+
+ // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
+ ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
+ "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
+ __func__,
+ streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
+ sessionId, transferType, uid, pid);
+
+ mThreadCanCallJava = threadCanCallJava;
+ mSelectedDeviceId = selectedDeviceId;
+ mSessionId = sessionId;
+
+ switch (transferType) {
+ case TRANSFER_DEFAULT:
+ if (sharedBuffer != 0) {
+ transferType = TRANSFER_SHARED;
+ } else if (cbf == NULL || threadCanCallJava) {
+ transferType = TRANSFER_SYNC;
+ } else {
+ transferType = TRANSFER_CALLBACK;
+ }
+ break;
+ case TRANSFER_CALLBACK:
+ case TRANSFER_SYNC_NOTIF_CALLBACK:
+ if (cbf == NULL || sharedBuffer != 0) {
+ ALOGE("%s(): Transfer type %s but cbf == NULL || sharedBuffer != 0",
+ convertTransferToText(transferType), __func__);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ break;
+ case TRANSFER_OBTAIN:
+ case TRANSFER_SYNC:
+ if (sharedBuffer != 0) {
+ ALOGE("%s(): Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ break;
+ case TRANSFER_SHARED:
+ if (sharedBuffer == 0) {
+ ALOGE("%s(): Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ break;
+ default:
+ ALOGE("%s(): Invalid transfer type %d",
+ __func__, transferType);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ mSharedBuffer = sharedBuffer;
+ mTransfer = transferType;
+ mDoNotReconnect = doNotReconnect;
+
+ ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
+ __func__, sharedBuffer->pointer(), sharedBuffer->size());
+
+ ALOGV("%s(): streamType %d frameCount %zu flags %04x",
+ __func__, streamType, frameCount, flags);
+
+ // invariant that mAudioTrack != 0 is true only after set() returns successfully
+ if (mAudioTrack != 0) {
+ ALOGE("%s(): Track already in use", __func__);
+ status = INVALID_OPERATION;
+ goto exit;
+ }
+
+ // handle default values first.
+ if (streamType == AUDIO_STREAM_DEFAULT) {
+ streamType = AUDIO_STREAM_MUSIC;
+ }
+ if (pAttributes == NULL) {
+ if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
+ ALOGE("%s(): Invalid stream type %d", __func__, streamType);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ mStreamType = streamType;
+
+ } else {
+ // stream type shouldn't be looked at, this track has audio attributes
+ memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
+ ALOGV("%s(): Building AudioTrack with attributes:"
+ " usage=%d content=%d flags=0x%x tags=[%s]",
+ __func__,
+ mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
+ mStreamType = AUDIO_STREAM_DEFAULT;
+ audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
+ }
+
+ // these below should probably come from the audioFlinger too...
+ if (format == AUDIO_FORMAT_DEFAULT) {
+ format = AUDIO_FORMAT_PCM_16_BIT;
+ } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
+ mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO;
+ }
+
+ // validate parameters
+ if (!audio_is_valid_format(format)) {
+ ALOGE("%s(): Invalid format %#x", __func__, format);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ mFormat = format;
+
+ if (!audio_is_output_channel(channelMask)) {
+ ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ mChannelMask = channelMask;
+ channelCount = audio_channel_count_from_out_mask(channelMask);
+ mChannelCount = channelCount;
+
+ // force direct flag if format is not linear PCM
+ // or offload was requested
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
+ || !audio_is_linear_pcm(format)) {
+ ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
+ ? "%s(): Offload request, forcing to Direct Output"
+ : "%s(): Not linear PCM, forcing to Direct Output",
+ __func__);
+ flags = (audio_output_flags_t)
+ // FIXME why can't we allow direct AND fast?
+ ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
+ }
+
+ // force direct flag if HW A/V sync requested
+ if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+
+ if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (audio_has_proportional_frames(format)) {
+ mFrameSize = channelCount * audio_bytes_per_sample(format);
+ } else {
+ mFrameSize = sizeof(uint8_t);
+ }
+ } else {
+ ALOG_ASSERT(audio_has_proportional_frames(format));
+ mFrameSize = channelCount * audio_bytes_per_sample(format);
+ // createTrack will return an error if PCM format is not supported by server,
+ // so no need to check for specific PCM formats here
+ }
+
+ // sampling rate must be specified for direct outputs
+ if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+ status = BAD_VALUE;
+ goto exit;
+ }
+ mSampleRate = sampleRate;
+ mOriginalSampleRate = sampleRate;
+ mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
+ mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
+
+ // Make copy of input parameter offloadInfo so that in the future:
+ // (a) createTrack_l doesn't need it as an input parameter
+ // (b) we can support re-creation of offloaded tracks
+ if (offloadInfo != NULL) {
+ mOffloadInfoCopy = *offloadInfo;
+ mOffloadInfo = &mOffloadInfoCopy;
+ } else {
+ mOffloadInfo = NULL;
+ memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
+ }
+
+ mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
+ mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
+ mSendLevel = 0.0f;
+ // mFrameCount is initialized in createTrack_l
+ mReqFrameCount = frameCount;
+ if (notificationFrames >= 0) {
+ mNotificationFramesReq = notificationFrames;
+ mNotificationsPerBufferReq = 0;
+ } else {
+ if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
+ ALOGE("%s(): notificationFrames=%d not permitted for non-fast track",
+ __func__, notificationFrames);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ if (frameCount > 0) {
+ ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
+ __func__, notificationFrames, frameCount);
+ status = BAD_VALUE;
+ goto exit;
+ }
+ mNotificationFramesReq = 0;
+ const uint32_t minNotificationsPerBuffer = 1;
+ const uint32_t maxNotificationsPerBuffer = 8;
+ mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
+ max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
+ ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
+ "%s(): notificationFrames=%d clamped to the range -%u to -%u",
+ __func__,
+ notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
+ }
+ mNotificationFramesAct = 0;
+ callingPid = IPCThreadState::self()->getCallingPid();
+ myPid = getpid();
+ if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
+ mClientUid = IPCThreadState::self()->getCallingUid();
+ } else {
+ mClientUid = uid;
+ }
+ if (pid == -1 || (callingPid != myPid)) {
+ mClientPid = callingPid;
+ } else {
+ mClientPid = pid;
+ }
+ mAuxEffectId = 0;
+ mOrigFlags = mFlags = flags;
+ mCbf = cbf;
+
+ if (cbf != NULL) {
+ mAudioTrackThread = new AudioTrackThread(*this);
+ mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
+ // thread begins in paused state, and will not reference us until start()
+ }
+
+ // create the IAudioTrack
+ {
+ AutoMutex lock(mLock);
+ status = createTrack_l();
+ }
+ if (status != NO_ERROR) {
+ if (mAudioTrackThread != 0) {
+ mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
+ mAudioTrackThread->requestExitAndWait();
+ mAudioTrackThread.clear();
+ }
+ goto exit;
+ }
+
+ mUserData = user;
+ mLoopCount = 0;
+ mLoopStart = 0;
+ mLoopEnd = 0;
+ mLoopCountNotified = 0;
+ mMarkerPosition = 0;
+ mMarkerReached = false;
+ mNewPosition = 0;
+ mUpdatePeriod = 0;
+ mPosition = 0;
+ mReleased = 0;
+ mStartNs = 0;
+ mStartFromZeroUs = 0;
+ AudioSystem::acquireAudioSessionId(mSessionId, mClientPid);
+ mSequence = 1;
+ mObservedSequence = mSequence;
+ mInUnderrun = false;
+ mPreviousTimestampValid = false;
+ mTimestampStartupGlitchReported = false;
+ mTimestampRetrogradePositionReported = false;
+ mTimestampRetrogradeTimeReported = false;
+ mTimestampStallReported = false;
+ mTimestampStaleTimeReported = false;
+ mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
+ mStartTs.mPosition = 0;
+ mUnderrunCountOffset = 0;
+ mFramesWritten = 0;
+ mFramesWrittenServerOffset = 0;
+ mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
+ mVolumeHandler = new media::VolumeHandler();
+
+exit:
+ mStatus = status;
+ return status;
+}
+
+// -------------------------------------------------------------------------
+
+status_t AudioTrack::start()
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
+
+ if (mState == STATE_ACTIVE) {
+ return INVALID_OPERATION;
+ }
+
+ mInUnderrun = true;
+
+ State previousState = mState;
+ if (previousState == STATE_PAUSED_STOPPING) {
+ mState = STATE_STOPPING;
+ } else {
+ mState = STATE_ACTIVE;
+ }
+ (void) updateAndGetPosition_l();
+
+ // save start timestamp
+ if (isOffloadedOrDirect_l()) {
+ if (getTimestamp_l(mStartTs) != OK) {
+ mStartTs.mPosition = 0;
+ }
+ } else {
+ if (getTimestamp_l(&mStartEts) != OK) {
+ mStartEts.clear();
+ }
+ }
+ mStartNs = systemTime(); // save this for timestamp adjustment after starting.
+ if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
+ // reset current position as seen by client to 0
+ mPosition = 0;
+ mPreviousTimestampValid = false;
+ mTimestampStartupGlitchReported = false;
+ mTimestampRetrogradePositionReported = false;
+ mTimestampRetrogradeTimeReported = false;
+ mTimestampStallReported = false;
+ mTimestampStaleTimeReported = false;
+ mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
+
+ if (!isOffloadedOrDirect_l()
+ && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
+ // Server side has consumed something, but is it finished consuming?
+ // It is possible since flush and stop are asynchronous that the server
+ // is still active at this point.
+ ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
+ __func__, mPortId,
+ (long long)(mFramesWrittenServerOffset
+ + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
+ (long long)mStartEts.mFlushed,
+ (long long)mFramesWritten);
+ // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
+ mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
+ }
+ mFramesWritten = 0;
+ mProxy->clearTimestamp(); // need new server push for valid timestamp
+ mMarkerReached = false;
+
+ // For offloaded tracks, we don't know if the hardware counters are really zero here,
+ // since the flush is asynchronous and stop may not fully drain.
+ // We save the time when the track is started to later verify whether
+ // the counters are realistic (i.e. start from zero after this time).
+ mStartFromZeroUs = mStartNs / 1000;
+
+ // force refresh of remaining frames by processAudioBuffer() as last
+ // write before stop could be partial.
+ mRefreshRemaining = true;
+
+ // for static track, clear the old flags when starting from stopped state
+ if (mSharedBuffer != 0) {
+ android_atomic_and(
+ ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
+ &mCblk->mFlags);
+ }
+ }
+ mNewPosition = mPosition + mUpdatePeriod;
+ int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
+
+ status_t status = NO_ERROR;
+ if (!(flags & CBLK_INVALID)) {
+ status = mAudioTrack->start();
+ if (status == DEAD_OBJECT) {
+ flags |= CBLK_INVALID;
+ }
+ }
+ if (flags & CBLK_INVALID) {
+ status = restoreTrack_l("start");
+ }
+
+ // resume or pause the callback thread as needed.
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (status == NO_ERROR) {
+ if (t != 0) {
+ if (previousState == STATE_STOPPING) {
+ mProxy->interrupt();
+ } else {
+ t->resume();
+ }
+ } else {
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ get_sched_policy(0, &mPreviousSchedulingGroup);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
+ }
+
+ // Start our local VolumeHandler for restoration purposes.
+ mVolumeHandler->setStarted();
+ } else {
+ ALOGE("%s(%d): status %d", __func__, mPortId, status);
+ mState = previousState;
+ if (t != 0) {
+ if (previousState != STATE_STOPPING) {
+ t->pause();
+ }
+ } else {
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ set_sched_policy(0, mPreviousSchedulingGroup);
+ }
+ }
+
+ return status;
+}
+
+void AudioTrack::stop()
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
+
+ if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
+ return;
+ }
+
+ if (isOffloaded_l()) {
+ mState = STATE_STOPPING;
+ } else {
+ mState = STATE_STOPPED;
+ ALOGD_IF(mSharedBuffer == nullptr,
+ "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
+ mReleased = 0;
+ }
+
+ mProxy->stop(); // notify server not to read beyond current client position until start().
+ mProxy->interrupt();
+ mAudioTrack->stop();
+
+ // Note: legacy handling - stop does not clear playback marker
+ // and periodic update counter, but flush does for streaming tracks.
+
+ if (mSharedBuffer != 0) {
+ // clear buffer position and loop count.
+ mStaticProxy->setBufferPositionAndLoop(0 /* position */,
+ 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
+ }
+
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ if (!isOffloaded_l()) {
+ t->pause();
+ } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
+ // causes wake up of the playback thread, that will callback the client for
+ // EVENT_STREAM_END in processAudioBuffer()
+ t->wake();
+ }
+ } else {
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ set_sched_policy(0, mPreviousSchedulingGroup);
+ }
+}
+
+bool AudioTrack::stopped() const
+{
+ AutoMutex lock(mLock);
+ return mState != STATE_ACTIVE;
+}
+
+void AudioTrack::flush()
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
+
+ if (mSharedBuffer != 0) {
+ return;
+ }
+ if (mState == STATE_ACTIVE) {
+ return;
+ }
+ flush_l();
+}
+
+void AudioTrack::flush_l()
+{
+ ALOG_ASSERT(mState != STATE_ACTIVE);
+
+ // clear playback marker and periodic update counter
+ mMarkerPosition = 0;
+ mMarkerReached = false;
+ mUpdatePeriod = 0;
+ mRefreshRemaining = true;
+
+ mState = STATE_FLUSHED;
+ mReleased = 0;
+ if (isOffloaded_l()) {
+ mProxy->interrupt();
+ }
+ mProxy->flush();
+ mAudioTrack->flush();
+}
+
+void AudioTrack::pause()
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
+
+ if (mState == STATE_ACTIVE) {
+ mState = STATE_PAUSED;
+ } else if (mState == STATE_STOPPING) {
+ mState = STATE_PAUSED_STOPPING;
+ } else {
+ return;
+ }
+ mProxy->interrupt();
+ mAudioTrack->pause();
+
+ if (isOffloaded_l()) {
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ // An offload output can be re-used between two audio tracks having
+ // the same configuration. A timestamp query for a paused track
+ // while the other is running would return an incorrect time.
+ // To fix this, cache the playback position on a pause() and return
+ // this time when requested until the track is resumed.
+
+ // OffloadThread sends HAL pause in its threadLoop. Time saved
+ // here can be slightly off.
+
+ // TODO: check return code for getRenderPosition.
+
+ uint32_t halFrames;
+ AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
+ ALOGV("%s(%d): for offload, cache current position %u",
+ __func__, mPortId, mPausedPosition);
+ }
+ }
+}
+
+status_t AudioTrack::setVolume(float left, float right)
+{
+ // This duplicates a test by AudioTrack JNI, but that is not the only caller
+ if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
+ isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ mVolume[AUDIO_INTERLEAVE_LEFT] = left;
+ mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
+
+ mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
+
+ if (isOffloaded_l()) {
+ mAudioTrack->signal();
+ }
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setVolume(float volume)
+{
+ return setVolume(volume, volume);
+}
+
+status_t AudioTrack::setAuxEffectSendLevel(float level)
+{
+ // This duplicates a test by AudioTrack JNI, but that is not the only caller
+ if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ mSendLevel = level;
+ mProxy->setSendLevel(level);
+
+ return NO_ERROR;
+}
+
+void AudioTrack::getAuxEffectSendLevel(float* level) const
+{
+ if (level != NULL) {
+ *level = mSendLevel;
+ }
+}
+
+status_t AudioTrack::setSampleRate(uint32_t rate)
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
+
+ if (rate == mSampleRate) {
+ return NO_ERROR;
+ }
+ if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
+ || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
+ return INVALID_OPERATION;
+ }
+ if (mOutput == AUDIO_IO_HANDLE_NONE) {
+ return NO_INIT;
+ }
+ // NOTE: it is theoretically possible, but highly unlikely, that a device change
+ // could mean a previously allowed sampling rate is no longer allowed.
+ uint32_t afSamplingRate;
+ if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
+ return NO_INIT;
+ }
+ // pitch is emulated by adjusting speed and sampleRate
+ const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
+ if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
+ return BAD_VALUE;
+ }
+ // TODO: Should we also check if the buffer size is compatible?
+
+ mSampleRate = rate;
+ mProxy->setSampleRate(effectiveSampleRate);
+
+ return NO_ERROR;
+}
+
+uint32_t AudioTrack::getSampleRate() const
+{
+ AutoMutex lock(mLock);
+
+ // sample rate can be updated during playback by the offloaded decoder so we need to
+ // query the HAL and update if needed.
+// FIXME use Proxy return channel to update the rate from server and avoid polling here
+ if (isOffloadedOrDirect_l()) {
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ uint32_t sampleRate = 0;
+ status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
+ if (status == NO_ERROR) {
+ mSampleRate = sampleRate;
+ }
+ }
+ }
+ return mSampleRate;
+}
+
+uint32_t AudioTrack::getOriginalSampleRate() const
+{
+ return mOriginalSampleRate;
+}
+
+status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
+{
+ AutoMutex lock(mLock);
+ if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
+ return NO_ERROR;
+ }
+ if (isOffloadedOrDirect_l()) {
+ return INVALID_OPERATION;
+ }
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
+ __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
+ // pitch is emulated by adjusting speed and sampleRate
+ const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
+ const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
+ const float effectivePitch = adjustPitch(playbackRate.mPitch);
+ AudioPlaybackRate playbackRateTemp = playbackRate;
+ playbackRateTemp.mSpeed = effectiveSpeed;
+ playbackRateTemp.mPitch = effectivePitch;
+
+ ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
+ __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
+
+ if (!isAudioPlaybackRateValid(playbackRateTemp)) {
+ ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
+ __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
+ return BAD_VALUE;
+ }
+ // Check if the buffer size is compatible.
+ if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
+ ALOGW("%s(%d) (%f, %f) failed (buffer size)",
+ __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
+ return BAD_VALUE;
+ }
+
+ // Check resampler ratios are within bounds
+ if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
+ (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
+ ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
+ __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
+ return BAD_VALUE;
+ }
+
+ if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
+ ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
+ __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
+ return BAD_VALUE;
+ }
+ mPlaybackRate = playbackRate;
+ //set effective rates
+ mProxy->setPlaybackRate(playbackRateTemp);
+ mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
+ return NO_ERROR;
+}
+
+const AudioPlaybackRate& AudioTrack::getPlaybackRate() const
+{
+ AutoMutex lock(mLock);
+ return mPlaybackRate;
+}
+
+ssize_t AudioTrack::getBufferSizeInFrames()
+{
+ AutoMutex lock(mLock);
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ return (ssize_t) mProxy->getBufferSizeInFrames();
+}
+
+status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
+{
+ if (duration == nullptr) {
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
+ if (bufferSizeInFrames < 0) {
+ return (status_t)bufferSizeInFrames;
+ }
+ *duration = (int64_t)((double)bufferSizeInFrames * 1000000
+ / ((double)mSampleRate * mPlaybackRate.mSpeed));
+ return NO_ERROR;
+}
+
+ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
+{
+ AutoMutex lock(mLock);
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ // Reject if timed track or compressed audio.
+ if (!audio_is_linear_pcm(mFormat)) {
+ return INVALID_OPERATION;
+ }
+ return (ssize_t) mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
+}
+
+status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
+{
+ if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
+ return INVALID_OPERATION;
+ }
+
+ if (loopCount == 0) {
+ ;
+ } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
+ loopEnd - loopStart >= MIN_LOOP) {
+ ;
+ } else {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ // See setPosition() regarding setting parameters such as loop points or position while active
+ if (mState == STATE_ACTIVE) {
+ return INVALID_OPERATION;
+ }
+ setLoop_l(loopStart, loopEnd, loopCount);
+ return NO_ERROR;
+}
+
+void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
+{
+ // We do not update the periodic notification point.
+ // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
+ mLoopCount = loopCount;
+ mLoopEnd = loopEnd;
+ mLoopStart = loopStart;
+ mLoopCountNotified = loopCount;
+ mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
+
+ // Waking the AudioTrackThread is not needed as this cannot be called when active.
+}
+
+status_t AudioTrack::setMarkerPosition(uint32_t marker)
+{
+ // The only purpose of setting marker position is to get a callback
+ if (mCbf == NULL || isOffloadedOrDirect()) {
+ return INVALID_OPERATION;
+ }
+
+ AutoMutex lock(mLock);
+ mMarkerPosition = marker;
+ mMarkerReached = false;
+
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ t->wake();
+ }
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
+{
+ if (isOffloadedOrDirect()) {
+ return INVALID_OPERATION;
+ }
+ if (marker == NULL) {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ mMarkerPosition.getValue(marker);
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
+{
+ // The only purpose of setting position update period is to get a callback
+ if (mCbf == NULL || isOffloadedOrDirect()) {
+ return INVALID_OPERATION;
+ }
+
+ AutoMutex lock(mLock);
+ mNewPosition = updateAndGetPosition_l() + updatePeriod;
+ mUpdatePeriod = updatePeriod;
+
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ t->wake();
+ }
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
+{
+ if (isOffloadedOrDirect()) {
+ return INVALID_OPERATION;
+ }
+ if (updatePeriod == NULL) {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ *updatePeriod = mUpdatePeriod;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setPosition(uint32_t position)
+{
+ if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
+ return INVALID_OPERATION;
+ }
+ if (position > mFrameCount) {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ // Currently we require that the player is inactive before setting parameters such as position
+ // or loop points. Otherwise, there could be a race condition: the application could read the
+ // current position, compute a new position or loop parameters, and then set that position or
+ // loop parameters but it would do the "wrong" thing since the position has continued to advance
+ // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
+ // to specify how it wants to handle such scenarios.
+ if (mState == STATE_ACTIVE) {
+ return INVALID_OPERATION;
+ }
+ // After setting the position, use full update period before notification.
+ mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
+ mStaticProxy->setBufferPosition(position);
+
+ // Waking the AudioTrackThread is not needed as this cannot be called when active.
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getPosition(uint32_t *position)
+{
+ if (position == NULL) {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ // FIXME: offloaded and direct tracks call into the HAL for render positions
+ // for compressed/synced data; however, we use proxy position for pure linear pcm data
+ // as we do not know the capability of the HAL for pcm position support and standby.
+ // There may be some latency differences between the HAL position and the proxy position.
+ if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
+ uint32_t dspFrames = 0;
+
+ if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
+ ALOGV("%s(%d): called in paused state, return cached position %u",
+ __func__, mPortId, mPausedPosition);
+ *position = mPausedPosition;
+ return NO_ERROR;
+ }
+
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ uint32_t halFrames; // actually unused
+ (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
+ // FIXME: on getRenderPosition() error, we return OK with frame position 0.
+ }
+ // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
+ // due to hardware latency. We leave this behavior for now.
+ *position = dspFrames;
+ } else {
+ if (mCblk->mFlags & CBLK_INVALID) {
+ (void) restoreTrack_l("getPosition");
+ // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
+ // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
+ }
+
+ // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
+ *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
+ 0 : updateAndGetPosition_l().value();
+ }
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getBufferPosition(uint32_t *position)
+{
+ if (mSharedBuffer == 0) {
+ return INVALID_OPERATION;
+ }
+ if (position == NULL) {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ *position = mStaticProxy->getBufferPosition();
+ return NO_ERROR;
+}
+
+status_t AudioTrack::reload()
+{
+ if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
+ return INVALID_OPERATION;
+ }
+
+ AutoMutex lock(mLock);
+ // See setPosition() regarding setting parameters such as loop points or position while active
+ if (mState == STATE_ACTIVE) {
+ return INVALID_OPERATION;
+ }
+ mNewPosition = mUpdatePeriod;
+ (void) updateAndGetPosition_l();
+ mPosition = 0;
+ mPreviousTimestampValid = false;
+#if 0
+ // The documentation is not clear on the behavior of reload() and the restoration
+ // of loop count. Historically we have not restored loop count, start, end,
+ // but it makes sense if one desires to repeat playing a particular sound.
+ if (mLoopCount != 0) {
+ mLoopCountNotified = mLoopCount;
+ mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
+ }
+#endif
+ mStaticProxy->setBufferPosition(0);
+ return NO_ERROR;
+}
+
+audio_io_handle_t AudioTrack::getOutput() const
+{
+ AutoMutex lock(mLock);
+ return mOutput;
+}
+
+status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
+ AutoMutex lock(mLock);
+ if (mSelectedDeviceId != deviceId) {
+ mSelectedDeviceId = deviceId;
+ if (mStatus == NO_ERROR) {
+ android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
+ mProxy->interrupt();
+ }
+ }
+ return NO_ERROR;
+}
+
+audio_port_handle_t AudioTrack::getOutputDevice() {
+ AutoMutex lock(mLock);
+ return mSelectedDeviceId;
+}
+
+// must be called with mLock held
+void AudioTrack::updateRoutedDeviceId_l()
+{
+ // if the track is inactive, do not update actual device as the output stream maybe routed
+ // to a device not relevant to this client because of other active use cases.
+ if (mState != STATE_ACTIVE) {
+ return;
+ }
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
+ if (deviceId != AUDIO_PORT_HANDLE_NONE) {
+ mRoutedDeviceId = deviceId;
+ }
+ }
+}
+
+audio_port_handle_t AudioTrack::getRoutedDeviceId() {
+ AutoMutex lock(mLock);
+ updateRoutedDeviceId_l();
+ return mRoutedDeviceId;
+}
+
+status_t AudioTrack::attachAuxEffect(int effectId)
+{
+ AutoMutex lock(mLock);
+ status_t status = mAudioTrack->attachAuxEffect(effectId);
+ if (status == NO_ERROR) {
+ mAuxEffectId = effectId;
+ }
+ return status;
+}
+
+audio_stream_type_t AudioTrack::streamType() const
+{
+ if (mStreamType == AUDIO_STREAM_DEFAULT) {
+ return AudioSystem::attributesToStreamType(mAttributes);
+ }
+ return mStreamType;
+}
+
+uint32_t AudioTrack::latency()
+{
+ AutoMutex lock(mLock);
+ updateLatency_l();
+ return mLatency;
+}
+
+// -------------------------------------------------------------------------
+
+// must be called with mLock held
+void AudioTrack::updateLatency_l()
+{
+ status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
+ if (status != NO_ERROR) {
+ ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
+ } else {
+ // FIXME don't believe this lie
+ mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
+ }
+}
+
+// TODO Move this macro to a common header file for enum to string conversion in audio framework.
+#define MEDIA_CASE_ENUM(name) case name: return #name
+const char * AudioTrack::convertTransferToText(transfer_type transferType) {
+ switch (transferType) {
+ MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
+ MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
+ MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
+ MEDIA_CASE_ENUM(TRANSFER_SYNC);
+ MEDIA_CASE_ENUM(TRANSFER_SHARED);
+ MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
+ default:
+ return "UNRECOGNIZED";
+ }
+}
+
+status_t AudioTrack::createTrack_l()
+{
+ status_t status;
+ bool callbackAdded = false;
+
+ const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
+ if (audioFlinger == 0) {
+ ALOGE("%s(%d): Could not get audioflinger",
+ __func__, mPortId);
+ status = NO_INIT;
+ goto exit;
+ }
+
+ {
+ // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
+ // After fast request is denied, we will request again if IAudioTrack is re-created.
+ // Client can only express a preference for FAST. Server will perform additional tests.
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
+ // either of these use cases:
+ // use case 1: shared buffer
+ bool sharedBuffer = mSharedBuffer != 0;
+ bool transferAllowed =
+ // use case 2: callback transfer mode
+ (mTransfer == TRANSFER_CALLBACK) ||
+ // use case 3: obtain/release mode
+ (mTransfer == TRANSFER_OBTAIN) ||
+ // use case 4: synchronous write
+ ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
+ && mThreadCanCallJava);
+
+ bool fastAllowed = sharedBuffer || transferAllowed;
+ if (!fastAllowed) {
+ ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
+ " not shared buffer and transfer = %s",
+ __func__, mPortId,
+ convertTransferToText(mTransfer));
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
+ }
+ }
+
+ IAudioFlinger::CreateTrackInput input;
+ if (mStreamType != AUDIO_STREAM_DEFAULT) {
+ input.attr = AudioSystem::streamTypeToAttributes(mStreamType);
+ } else {
+ input.attr = mAttributes;
+ }
+ input.config = AUDIO_CONFIG_INITIALIZER;
+ input.config.sample_rate = mSampleRate;
+ input.config.channel_mask = mChannelMask;
+ input.config.format = mFormat;
+ input.config.offload_info = mOffloadInfoCopy;
+ input.clientInfo.clientUid = mClientUid;
+ input.clientInfo.clientPid = mClientPid;
+ input.clientInfo.clientTid = -1;
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
+ // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
+ // application-level code follows all non-blocking design rules, the language runtime
+ // doesn't also follow those rules, so the thread will not benefit overall.
+ if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
+ input.clientInfo.clientTid = mAudioTrackThread->getTid();
+ }
+ }
+ input.sharedBuffer = mSharedBuffer;
+ input.notificationsPerBuffer = mNotificationsPerBufferReq;
+ input.speed = 1.0;
+ if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
+ (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+ input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
+ max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
+ }
+ input.flags = mFlags;
+ input.frameCount = mReqFrameCount;
+ input.notificationFrameCount = mNotificationFramesReq;
+ input.selectedDeviceId = mSelectedDeviceId;
+ input.sessionId = mSessionId;
+
+ IAudioFlinger::CreateTrackOutput output;
+
+ sp<IAudioTrack> track = audioFlinger->createTrack(input,
+ output,
+ &status);
+
+ if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
+ ALOGE("%s(%d): AudioFlinger could not create track, status: %d output %d",
+ __func__, mPortId, status, output.outputId);
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ goto exit;
+ }
+ ALOG_ASSERT(track != 0);
+
+ mFrameCount = output.frameCount;
+ mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
+ mRoutedDeviceId = output.selectedDeviceId;
+ mSessionId = output.sessionId;
+
+ mSampleRate = output.sampleRate;
+ if (mOriginalSampleRate == 0) {
+ mOriginalSampleRate = mSampleRate;
+ }
+
+ mAfFrameCount = output.afFrameCount;
+ mAfSampleRate = output.afSampleRate;
+ mAfLatency = output.afLatencyMs;
+
+ mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
+
+ // AudioFlinger now owns the reference to the I/O handle,
+ // so we are no longer responsible for releasing it.
+
+ // FIXME compare to AudioRecord
+ sp<IMemory> iMem = track->getCblk();
+ if (iMem == 0) {
+ ALOGE("%s(%d): Could not get control block", __func__, mPortId);
+ status = NO_INIT;
+ goto exit;
+ }
+ void *iMemPointer = iMem->pointer();
+ if (iMemPointer == NULL) {
+ ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId);
+ status = NO_INIT;
+ goto exit;
+ }
+ // invariant that mAudioTrack != 0 is true only after set() returns successfully
+ if (mAudioTrack != 0) {
+ IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
+ mDeathNotifier.clear();
+ }
+ mAudioTrack = track;
+ mCblkMemory = iMem;
+ IPCThreadState::self()->flushCommands();
+
+ audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
+ mCblk = cblk;
+
+ mAwaitBoost = false;
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
+ if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
+ ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
+ __func__, mPortId, mReqFrameCount, mFrameCount);
+ if (!mThreadCanCallJava) {
+ mAwaitBoost = true;
+ }
+ } else {
+ ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
+ __func__, mPortId, mReqFrameCount, mFrameCount);
+ }
+ }
+ mFlags = output.flags;
+
+ //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
+ if (mDeviceCallback != 0) {
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
+ }
+ AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
+ callbackAdded = true;
+ }
+
+ mPortId = output.portId;
+ // We retain a copy of the I/O handle, but don't own the reference
+ mOutput = output.outputId;
+ mRefreshRemaining = true;
+
+ // Starting address of buffers in shared memory. If there is a shared buffer, buffers
+ // is the value of pointer() for the shared buffer, otherwise buffers points
+ // immediately after the control block. This address is for the mapping within client
+ // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
+ void* buffers;
+ if (mSharedBuffer == 0) {
+ buffers = cblk + 1;
+ } else {
+ buffers = mSharedBuffer->pointer();
+ if (buffers == NULL) {
+ ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
+ status = NO_INIT;
+ goto exit;
+ }
+ }
+
+ mAudioTrack->attachAuxEffect(mAuxEffectId);
+
+ // If IAudioTrack is re-created, don't let the requested frameCount
+ // decrease. This can confuse clients that cache frameCount().
+ if (mFrameCount > mReqFrameCount) {
+ mReqFrameCount = mFrameCount;
+ }
+
+ // reset server position to 0 as we have new cblk.
+ mServer = 0;
+
+ // update proxy
+ if (mSharedBuffer == 0) {
+ mStaticProxy.clear();
+ mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
+ } else {
+ mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
+ mProxy = mStaticProxy;
+ }
+
+ mProxy->setVolumeLR(gain_minifloat_pack(
+ gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
+ gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
+
+ mProxy->setSendLevel(mSendLevel);
+ const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
+ const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
+ const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
+ mProxy->setSampleRate(effectiveSampleRate);
+
+ AudioPlaybackRate playbackRateTemp = mPlaybackRate;
+ playbackRateTemp.mSpeed = effectiveSpeed;
+ playbackRateTemp.mPitch = effectivePitch;
+ mProxy->setPlaybackRate(playbackRateTemp);
+ mProxy->setMinimum(mNotificationFramesAct);
+
+ mDeathNotifier = new DeathNotifier(this);
+ IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
+
+ }
+
+exit:
+ if (status != NO_ERROR && callbackAdded) {
+ // note: mOutput is always valid is callbackAdded is true
+ AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
+ }
+
+ mStatus = status;
+
+ // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
+ return status;
+}
+
+status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
+{
+ if (audioBuffer == NULL) {
+ if (nonContig != NULL) {
+ *nonContig = 0;
+ }
+ return BAD_VALUE;
+ }
+ if (mTransfer != TRANSFER_OBTAIN) {
+ audioBuffer->frameCount = 0;
+ audioBuffer->size = 0;
+ audioBuffer->raw = NULL;
+ if (nonContig != NULL) {
+ *nonContig = 0;
+ }
+ return INVALID_OPERATION;
+ }
+
+ const struct timespec *requested;
+ struct timespec timeout;
+ if (waitCount == -1) {
+ requested = &ClientProxy::kForever;
+ } else if (waitCount == 0) {
+ requested = &ClientProxy::kNonBlocking;
+ } else if (waitCount > 0) {
+ time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
+ timeout.tv_sec = ms / 1000;
+ timeout.tv_nsec = (long) (ms % 1000) * 1000000;
+ requested = &timeout;
+ } else {
+ ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
+ requested = NULL;
+ }
+ return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
+}
+
+status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
+ struct timespec *elapsed, size_t *nonContig)
+{
+ // previous and new IAudioTrack sequence numbers are used to detect track re-creation
+ uint32_t oldSequence = 0;
+ uint32_t newSequence;
+
+ Proxy::Buffer buffer;
+ status_t status = NO_ERROR;
+
+ static const int32_t kMaxTries = 5;
+ int32_t tryCounter = kMaxTries;
+
+ do {
+ // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
+ // keep them from going away if another thread re-creates the track during obtainBuffer()
+ sp<AudioTrackClientProxy> proxy;
+ sp<IMemory> iMem;
+
+ { // start of lock scope
+ AutoMutex lock(mLock);
+
+ newSequence = mSequence;
+ // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
+ if (status == DEAD_OBJECT) {
+ // re-create track, unless someone else has already done so
+ if (newSequence == oldSequence) {
+ status = restoreTrack_l("obtainBuffer");
+ if (status != NO_ERROR) {
+ buffer.mFrameCount = 0;
+ buffer.mRaw = NULL;
+ buffer.mNonContig = 0;
+ break;
+ }
+ }
+ }
+ oldSequence = newSequence;
+
+ if (status == NOT_ENOUGH_DATA) {
+ restartIfDisabled();
+ }
+
+ // Keep the extra references
+ proxy = mProxy;
+ iMem = mCblkMemory;
+
+ if (mState == STATE_STOPPING) {
+ status = -EINTR;
+ buffer.mFrameCount = 0;
+ buffer.mRaw = NULL;
+ buffer.mNonContig = 0;
+ break;
+ }
+
+ // Non-blocking if track is stopped or paused
+ if (mState != STATE_ACTIVE) {
+ requested = &ClientProxy::kNonBlocking;
+ }
+
+ } // end of lock scope
+
+ buffer.mFrameCount = audioBuffer->frameCount;
+ // FIXME starts the requested timeout and elapsed over from scratch
+ status = proxy->obtainBuffer(&buffer, requested, elapsed);
+ } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
+
+ audioBuffer->frameCount = buffer.mFrameCount;
+ audioBuffer->size = buffer.mFrameCount * mFrameSize;
+ audioBuffer->raw = buffer.mRaw;
+ if (nonContig != NULL) {
+ *nonContig = buffer.mNonContig;
+ }
+ return status;
+}
+
+void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
+{
+ // FIXME add error checking on mode, by adding an internal version
+ if (mTransfer == TRANSFER_SHARED) {
+ return;
+ }
+
+ size_t stepCount = audioBuffer->size / mFrameSize;
+ if (stepCount == 0) {
+ return;
+ }
+
+ Proxy::Buffer buffer;
+ buffer.mFrameCount = stepCount;
+ buffer.mRaw = audioBuffer->raw;
+
+ AutoMutex lock(mLock);
+ mReleased += stepCount;
+ mInUnderrun = false;
+ mProxy->releaseBuffer(&buffer);
+
+ // restart track if it was disabled by audioflinger due to previous underrun
+ restartIfDisabled();
+}
+
+void AudioTrack::restartIfDisabled()
+{
+ int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
+ if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
+ ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
+ __func__, mPortId, this);
+ // FIXME ignoring status
+ mAudioTrack->start();
+ }
+}
+
+// -------------------------------------------------------------------------
+
+ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
+{
+ if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
+ return INVALID_OPERATION;
+ }
+
+ if (isDirect()) {
+ AutoMutex lock(mLock);
+ int32_t flags = android_atomic_and(
+ ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
+ &mCblk->mFlags);
+ if (flags & CBLK_INVALID) {
+ return DEAD_OBJECT;
+ }
+ }
+
+ if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
+ // Validity-check: user is most-likely passing an error code, and it would
+ // make the return value ambiguous (actualSize vs error).
+ ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
+ __func__, mPortId, buffer, userSize, userSize);
+ return BAD_VALUE;
+ }
+
+ size_t written = 0;
+ Buffer audioBuffer;
+
+ while (userSize >= mFrameSize) {
+ audioBuffer.frameCount = userSize / mFrameSize;
+
+ status_t err = obtainBuffer(&audioBuffer,
+ blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
+ if (err < 0) {
+ if (written > 0) {
+ break;
+ }
+ if (err == TIMED_OUT || err == -EINTR) {
+ err = WOULD_BLOCK;
+ }
+ return ssize_t(err);
+ }
+
+ size_t toWrite = audioBuffer.size;
+ memcpy(audioBuffer.i8, buffer, toWrite);
+ buffer = ((const char *) buffer) + toWrite;
+ userSize -= toWrite;
+ written += toWrite;
+
+ releaseBuffer(&audioBuffer);
+ }
+
+ if (written > 0) {
+ mFramesWritten += written / mFrameSize;
+
+ if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
+ const sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ // causes wake up of the playback thread, that will callback the client for
+ // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
+ t->wake();
+ }
+ }
+ }
+
+ return written;
+}
+
+// -------------------------------------------------------------------------
+
+nsecs_t AudioTrack::processAudioBuffer()
+{
+ // Currently the AudioTrack thread is not created if there are no callbacks.
+ // Would it ever make sense to run the thread, even without callbacks?
+ // If so, then replace this by checks at each use for mCbf != NULL.
+ LOG_ALWAYS_FATAL_IF(mCblk == NULL);
+
+ mLock.lock();
+ if (mAwaitBoost) {
+ mAwaitBoost = false;
+ mLock.unlock();
+ static const int32_t kMaxTries = 5;
+ int32_t tryCounter = kMaxTries;
+ uint32_t pollUs = 10000;
+ do {
+ int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
+ if (policy == SCHED_FIFO || policy == SCHED_RR) {
+ break;
+ }
+ usleep(pollUs);
+ pollUs <<= 1;
+ } while (tryCounter-- > 0);
+ if (tryCounter < 0) {
+ ALOGE("%s(%d): did not receive expected priority boost on time",
+ __func__, mPortId);
+ }
+ // Run again immediately
+ return 0;
+ }
+
+ // Can only reference mCblk while locked
+ int32_t flags = android_atomic_and(
+ ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
+
+ // Check for track invalidation
+ if (flags & CBLK_INVALID) {
+ // for offloaded tracks restoreTrack_l() will just update the sequence and clear
+ // AudioSystem cache. We should not exit here but after calling the callback so
+ // that the upper layers can recreate the track
+ if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
+ status_t status __unused = restoreTrack_l("processAudioBuffer");
+ // FIXME unused status
+ // after restoration, continue below to make sure that the loop and buffer events
+ // are notified because they have been cleared from mCblk->mFlags above.
+ }
+ }
+
+ bool waitStreamEnd = mState == STATE_STOPPING;
+ bool active = mState == STATE_ACTIVE;
+
+ // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
+ bool newUnderrun = false;
+ if (flags & CBLK_UNDERRUN) {
+#if 0
+ // Currently in shared buffer mode, when the server reaches the end of buffer,
+ // the track stays active in continuous underrun state. It's up to the application
+ // to pause or stop the track, or set the position to a new offset within buffer.
+ // This was some experimental code to auto-pause on underrun. Keeping it here
+ // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
+ if (mTransfer == TRANSFER_SHARED) {
+ mState = STATE_PAUSED;
+ active = false;
+ }
+#endif
+ if (!mInUnderrun) {
+ mInUnderrun = true;
+ newUnderrun = true;
+ }
+ }
+
+ // Get current position of server
+ Modulo<uint32_t> position(updateAndGetPosition_l());
+
+ // Manage marker callback
+ bool markerReached = false;
+ Modulo<uint32_t> markerPosition(mMarkerPosition);
+ // uses 32 bit wraparound for comparison with position.
+ if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
+ mMarkerReached = markerReached = true;
+ }
+
+ // Determine number of new position callback(s) that will be needed, while locked
+ size_t newPosCount = 0;
+ Modulo<uint32_t> newPosition(mNewPosition);
+ uint32_t updatePeriod = mUpdatePeriod;
+ // FIXME fails for wraparound, need 64 bits
+ if (updatePeriod > 0 && position >= newPosition) {
+ newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
+ mNewPosition += updatePeriod * newPosCount;
+ }
+
+ // Cache other fields that will be needed soon
+ uint32_t sampleRate = mSampleRate;
+ float speed = mPlaybackRate.mSpeed;
+ const uint32_t notificationFrames = mNotificationFramesAct;
+ if (mRefreshRemaining) {
+ mRefreshRemaining = false;
+ mRemainingFrames = notificationFrames;
+ mRetryOnPartialBuffer = false;
+ }
+ size_t misalignment = mProxy->getMisalignment();
+ uint32_t sequence = mSequence;
+ sp<AudioTrackClientProxy> proxy = mProxy;
+
+ // Determine the number of new loop callback(s) that will be needed, while locked.
+ int loopCountNotifications = 0;
+ uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
+
+ if (mLoopCount > 0) {
+ int loopCount;
+ size_t bufferPosition;
+ mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
+ loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
+ loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
+ mLoopCountNotified = loopCount; // discard any excess notifications
+ } else if (mLoopCount < 0) {
+ // FIXME: We're not accurate with notification count and position with infinite looping
+ // since loopCount from server side will always return -1 (we could decrement it).
+ size_t bufferPosition = mStaticProxy->getBufferPosition();
+ loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
+ loopPeriod = mLoopEnd - bufferPosition;
+ } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
+ size_t bufferPosition = mStaticProxy->getBufferPosition();
+ loopPeriod = mFrameCount - bufferPosition;
+ }
+
+ // These fields don't need to be cached, because they are assigned only by set():
+ // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags
+ // mFlags is also assigned by createTrack_l(), but not the bit we care about.
+
+ mLock.unlock();
+
+ // get anchor time to account for callbacks.
+ const nsecs_t timeBeforeCallbacks = systemTime();
+
+ if (waitStreamEnd) {
+ // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
+ // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
+ // (and make sure we don't callback for more data while we're stopping).
+ // This helps with position, marker notifications, and track invalidation.
+ struct timespec timeout;
+ timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
+ timeout.tv_nsec = 0;
+
+ status_t status = proxy->waitStreamEndDone(&timeout);
+ switch (status) {
+ case NO_ERROR:
+ case DEAD_OBJECT:
+ case TIMED_OUT:
+ if (status != DEAD_OBJECT) {
+ // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
+ // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
+ mCbf(EVENT_STREAM_END, mUserData, NULL);
+ }
+ {
+ AutoMutex lock(mLock);
+ // The previously assigned value of waitStreamEnd is no longer valid,
+ // since the mutex has been unlocked and either the callback handler
+ // or another thread could have re-started the AudioTrack during that time.
+ waitStreamEnd = mState == STATE_STOPPING;
+ if (waitStreamEnd) {
+ mState = STATE_STOPPED;
+ mReleased = 0;
+ }
+ }
+ if (waitStreamEnd && status != DEAD_OBJECT) {
+ return NS_INACTIVE;
+ }
+ break;
+ }
+ return 0;
+ }
+
+ // perform callbacks while unlocked
+ if (newUnderrun) {
+ mCbf(EVENT_UNDERRUN, mUserData, NULL);
+ }
+ while (loopCountNotifications > 0) {
+ mCbf(EVENT_LOOP_END, mUserData, NULL);
+ --loopCountNotifications;
+ }
+ if (flags & CBLK_BUFFER_END) {
+ mCbf(EVENT_BUFFER_END, mUserData, NULL);
+ }
+ if (markerReached) {
+ mCbf(EVENT_MARKER, mUserData, &markerPosition);
+ }
+ while (newPosCount > 0) {
+ size_t temp = newPosition.value(); // FIXME size_t != uint32_t
+ mCbf(EVENT_NEW_POS, mUserData, &temp);
+ newPosition += updatePeriod;
+ newPosCount--;
+ }
+
+ if (mObservedSequence != sequence) {
+ mObservedSequence = sequence;
+ mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
+ // for offloaded tracks, just wait for the upper layers to recreate the track
+ if (isOffloadedOrDirect()) {
+ return NS_INACTIVE;
+ }
+ }
+
+ // if inactive, then don't run me again until re-started
+ if (!active) {
+ return NS_INACTIVE;
+ }
+
+ // Compute the estimated time until the next timed event (position, markers, loops)
+ // FIXME only for non-compressed audio
+ uint32_t minFrames = ~0;
+ if (!markerReached && position < markerPosition) {
+ minFrames = (markerPosition - position).value();
+ }
+ if (loopPeriod > 0 && loopPeriod < minFrames) {
+ // loopPeriod is already adjusted for actual position.
+ minFrames = loopPeriod;
+ }
+ if (updatePeriod > 0) {
+ minFrames = min(minFrames, (newPosition - position).value());
+ }
+
+ // If > 0, poll periodically to recover from a stuck server. A good value is 2.
+ static const uint32_t kPoll = 0;
+ if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
+ minFrames = kPoll * notificationFrames;
+ }
+
+ // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
+ static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
+ const nsecs_t timeAfterCallbacks = systemTime();
+
+ // Convert frame units to time units
+ nsecs_t ns = NS_WHENEVER;
+ if (minFrames != (uint32_t) ~0) {
+ // AudioFlinger consumption of client data may be irregular when coming out of device
+ // standby since the kernel buffers require filling. This is throttled to no more than 2x
+ // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
+ // half (but no more than half a second) to improve callback accuracy during these temporary
+ // data surges.
+ const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
+ constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
+ ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
+ ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
+ // TODO: Should we warn if the callback time is too long?
+ if (ns < 0) ns = 0;
+ }
+
+ // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
+ if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
+ return ns;
+ }
+
+ // EVENT_MORE_DATA callback handling.
+ // Timing for linear pcm audio data formats can be derived directly from the
+ // buffer fill level.
+ // Timing for compressed data is not directly available from the buffer fill level,
+ // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
+ // to return a certain fill level.
+
+ struct timespec timeout;
+ const struct timespec *requested = &ClientProxy::kForever;
+ if (ns != NS_WHENEVER) {
+ timeout.tv_sec = ns / 1000000000LL;
+ timeout.tv_nsec = ns % 1000000000LL;
+ ALOGV("%s(%d): timeout %ld.%03d",
+ __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
+ requested = &timeout;
+ }
+
+ size_t writtenFrames = 0;
+ while (mRemainingFrames > 0) {
+
+ Buffer audioBuffer;
+ audioBuffer.frameCount = mRemainingFrames;
+ size_t nonContig;
+ status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
+ LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
+ "%s(%d): obtainBuffer() err=%d frameCount=%zu",
+ __func__, mPortId, err, audioBuffer.frameCount);
+ requested = &ClientProxy::kNonBlocking;
+ size_t avail = audioBuffer.frameCount + nonContig;
+ ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
+ __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
+ if (err != NO_ERROR) {
+ if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
+ (isOffloaded() && (err == DEAD_OBJECT))) {
+ // FIXME bug 25195759
+ return 1000000;
+ }
+ ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
+ __func__, mPortId, err);
+ return NS_NEVER;
+ }
+
+ if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
+ mRetryOnPartialBuffer = false;
+ if (avail < mRemainingFrames) {
+ if (ns > 0) { // account for obtain time
+ const nsecs_t timeNow = systemTime();
+ ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
+ }
+
+ // delayNs is first computed by the additional frames required in the buffer.
+ nsecs_t delayNs = framesToNanoseconds(
+ mRemainingFrames - avail, sampleRate, speed);
+
+ // afNs is the AudioFlinger mixer period in ns.
+ const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
+
+ // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
+ // we may have a race if we wait based on the number of frames desired.
+ // This is a possible issue with resampling and AAudio.
+ //
+ // The granularity of audioflinger processing is one mixer period; if
+ // our wait time is less than one mixer period, wait at most half the period.
+ if (delayNs < afNs) {
+ delayNs = std::min(delayNs, afNs / 2);
+ }
+
+ // adjust our ns wait by delayNs.
+ if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
+ ns = delayNs;
+ }
+ return ns;
+ }
+ }
+
+ size_t reqSize = audioBuffer.size;
+ if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
+ // when notifying client it can write more data, pass the total size that can be
+ // written in the next write() call, since it's not passed through the callback
+ audioBuffer.size += nonContig;
+ }
+ mCbf(mTransfer == TRANSFER_CALLBACK ? EVENT_MORE_DATA : EVENT_CAN_WRITE_MORE_DATA,
+ mUserData, &audioBuffer);
+ size_t writtenSize = audioBuffer.size;
+
+ // Validity check on returned size
+ if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
+ ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
+ __func__, mPortId, reqSize, ssize_t(writtenSize));
+ return NS_NEVER;
+ }
+
+ if (writtenSize == 0) {
+ if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
+ // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
+ // android.media.AudioTrack. The JNI is not using the callback to provide data,
+ // it only signals to the Java client that it can provide more data, which
+ // this track is read to accept now.
+ // The playback thread will be awaken at the next ::write()
+ return NS_WHENEVER;
+ }
+ // The callback is done filling buffers
+ // Keep this thread going to handle timed events and
+ // still try to get more data in intervals of WAIT_PERIOD_MS
+ // but don't just loop and block the CPU, so wait
+
+ // mCbf(EVENT_MORE_DATA, ...) might either
+ // (1) Block until it can fill the buffer, returning 0 size on EOS.
+ // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
+ // (3) Return 0 size when no data is available, does not wait for more data.
+ //
+ // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
+ // We try to compute the wait time to avoid a tight sleep-wait cycle,
+ // especially for case (3).
+ //
+ // The decision to support (1) and (2) affect the sizing of mRemainingFrames
+ // and this loop; whereas for case (3) we could simply check once with the full
+ // buffer size and skip the loop entirely.
+
+ nsecs_t myns;
+ if (audio_has_proportional_frames(mFormat)) {
+ // time to wait based on buffer occupancy
+ const nsecs_t datans = mRemainingFrames <= avail ? 0 :
+ framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
+ // audio flinger thread buffer size (TODO: adjust for fast tracks)
+ // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
+ const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
+ // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
+ myns = datans + (afns / 2);
+ } else {
+ // FIXME: This could ping quite a bit if the buffer isn't full.
+ // Note that when mState is stopping we waitStreamEnd, so it never gets here.
+ myns = kWaitPeriodNs;
+ }
+ if (ns > 0) { // account for obtain and callback time
+ const nsecs_t timeNow = systemTime();
+ ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
+ }
+ if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
+ ns = myns;
+ }
+ return ns;
+ }
+
+ size_t releasedFrames = writtenSize / mFrameSize;
+ audioBuffer.frameCount = releasedFrames;
+ mRemainingFrames -= releasedFrames;
+ if (misalignment >= releasedFrames) {
+ misalignment -= releasedFrames;
+ } else {
+ misalignment = 0;
+ }
+
+ releaseBuffer(&audioBuffer);
+ writtenFrames += releasedFrames;
+
+ // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
+ // if callback doesn't like to accept the full chunk
+ if (writtenSize < reqSize) {
+ continue;
+ }
+
+ // There could be enough non-contiguous frames available to satisfy the remaining request
+ if (mRemainingFrames <= nonContig) {
+ continue;
+ }
+
+#if 0
+ // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
+ // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
+ // that total to a sum == notificationFrames.
+ if (0 < misalignment && misalignment <= mRemainingFrames) {
+ mRemainingFrames = misalignment;
+ return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
+ }
+#endif
+
+ }
+ if (writtenFrames > 0) {
+ AutoMutex lock(mLock);
+ mFramesWritten += writtenFrames;
+ }
+ mRemainingFrames = notificationFrames;
+ mRetryOnPartialBuffer = true;
+
+ // A lot has transpired since ns was calculated, so run again immediately and re-calculate
+ return 0;
+}
+
+status_t AudioTrack::restoreTrack_l(const char *from)
+{
+ ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
+ __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
+ ++mSequence;
+
+ // refresh the audio configuration cache in this process to make sure we get new
+ // output parameters and new IAudioFlinger in createTrack_l()
+ AudioSystem::clearAudioConfigCache();
+
+ if (isOffloadedOrDirect_l() || mDoNotReconnect) {
+ // FIXME re-creation of offloaded and direct tracks is not yet implemented;
+ // reconsider enabling for linear PCM encodings when position can be preserved.
+ return DEAD_OBJECT;
+ }
+
+ // Save so we can return count since creation.
+ mUnderrunCountOffset = getUnderrunCount_l();
+
+ // save the old static buffer position
+ uint32_t staticPosition = 0;
+ size_t bufferPosition = 0;
+ int loopCount = 0;
+ if (mStaticProxy != 0) {
+ mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
+ staticPosition = mStaticProxy->getPosition().unsignedValue();
+ }
+
+ // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
+ // causes a lot of churn on the service side, and it can reject starting
+ // playback of a previously created track. May also apply to other cases.
+ const int INITIAL_RETRIES = 3;
+ int retries = INITIAL_RETRIES;
+retry:
+ if (retries < INITIAL_RETRIES) {
+ // See the comment for clearAudioConfigCache at the start of the function.
+ AudioSystem::clearAudioConfigCache();
+ }
+ mFlags = mOrigFlags;
+
+ // If a new IAudioTrack is successfully created, createTrack_l() will modify the
+ // following member variables: mAudioTrack, mCblkMemory and mCblk.
+ // It will also delete the strong references on previous IAudioTrack and IMemory.
+ // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
+ status_t result = createTrack_l();
+
+ if (result == NO_ERROR) {
+ // take the frames that will be lost by track recreation into account in saved position
+ // For streaming tracks, this is the amount we obtained from the user/client
+ // (not the number actually consumed at the server - those are already lost).
+ if (mStaticProxy == 0) {
+ mPosition = mReleased;
+ }
+ // Continue playback from last known position and restore loop.
+ if (mStaticProxy != 0) {
+ if (loopCount != 0) {
+ mStaticProxy->setBufferPositionAndLoop(bufferPosition,
+ mLoopStart, mLoopEnd, loopCount);
+ } else {
+ mStaticProxy->setBufferPosition(bufferPosition);
+ if (bufferPosition == mFrameCount) {
+ ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
+ }
+ }
+ }
+ // restore volume handler
+ mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
+ sp<VolumeShaper::Operation> operationToEnd =
+ new VolumeShaper::Operation(shaper.mOperation);
+ // TODO: Ideally we would restore to the exact xOffset position
+ // as returned by getVolumeShaperState(), but we don't have that
+ // information when restoring at the client unless we periodically poll
+ // the server or create shared memory state.
+ //
+ // For now, we simply advance to the end of the VolumeShaper effect
+ // if it has been started.
+ if (shaper.isStarted()) {
+ operationToEnd->setNormalizedTime(1.f);
+ }
+ return mAudioTrack->applyVolumeShaper(shaper.mConfiguration, operationToEnd);
+ });
+
+ if (mState == STATE_ACTIVE) {
+ result = mAudioTrack->start();
+ }
+ // server resets to zero so we offset
+ mFramesWrittenServerOffset =
+ mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
+ mFramesWrittenAtRestore = mFramesWrittenServerOffset;
+ }
+ if (result != NO_ERROR) {
+ ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
+ if (--retries > 0) {
+ // leave time for an eventual race condition to clear before retrying
+ usleep(500000);
+ goto retry;
+ }
+ // if no retries left, set invalid bit to force restoring at next occasion
+ // and avoid inconsistent active state on client and server sides
+ if (mCblk != nullptr) {
+ android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
+ }
+ }
+ return result;
+}
+
+Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
+{
+ // This is the sole place to read server consumed frames
+ Modulo<uint32_t> newServer(mProxy->getPosition());
+ const int32_t delta = (newServer - mServer).signedValue();
+ // TODO There is controversy about whether there can be "negative jitter" in server position.
+ // This should be investigated further, and if possible, it should be addressed.
+ // A more definite failure mode is infrequent polling by client.
+ // One could call (void)getPosition_l() in releaseBuffer(),
+ // so mReleased and mPosition are always lock-step as best possible.
+ // That should ensure delta never goes negative for infrequent polling
+ // unless the server has more than 2^31 frames in its buffer,
+ // in which case the use of uint32_t for these counters has bigger issues.
+ ALOGE_IF(delta < 0,
+ "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
+ __func__, mPortId, delta);
+ mServer = newServer;
+ if (delta > 0) { // avoid retrograde
+ mPosition += delta;
+ }
+ return mPosition;
+}
+
+bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
+{
+ updateLatency_l();
+ // applicable for mixing tracks only (not offloaded or direct)
+ if (mStaticProxy != 0) {
+ return true; // static tracks do not have issues with buffer sizing.
+ }
+ const size_t minFrameCount =
+ AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
+ sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
+ const bool allowed = mFrameCount >= minFrameCount;
+ ALOGD_IF(!allowed,
+ "%s(%d): denied "
+ "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
+ "mFrameCount:%zu < minFrameCount:%zu",
+ __func__, mPortId,
+ mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
+ mFrameCount, minFrameCount);
+ return allowed;
+}
+
+status_t AudioTrack::setParameters(const String8& keyValuePairs)
+{
+ AutoMutex lock(mLock);
+ return mAudioTrack->setParameters(keyValuePairs);
+}
+
+status_t AudioTrack::selectPresentation(int presentationId, int programId)
+{
+ AutoMutex lock(mLock);
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
+ param.addInt(String8(AudioParameter::keyProgramId), programId);
+ ALOGV("%s(%d): PresentationId/ProgramId[%s]",
+ __func__, mPortId, param.toString().string());
+
+ return mAudioTrack->setParameters(param.toString());
+}
+
+VolumeShaper::Status AudioTrack::applyVolumeShaper(
+ const sp<VolumeShaper::Configuration>& configuration,
+ const sp<VolumeShaper::Operation>& operation)
+{
+ AutoMutex lock(mLock);
+ mVolumeHandler->setIdIfNecessary(configuration);
+ VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(configuration, operation);
+
+ if (status == DEAD_OBJECT) {
+ if (restoreTrack_l("applyVolumeShaper") == OK) {
+ status = mAudioTrack->applyVolumeShaper(configuration, operation);
+ }
+ }
+ if (status >= 0) {
+ // save VolumeShaper for restore
+ mVolumeHandler->applyVolumeShaper(configuration, operation);
+ if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
+ mVolumeHandler->setStarted();
+ }
+ } else {
+ // warn only if not an expected restore failure.
+ ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
+ "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
+ }
+ return status;
+}
+
+sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
+{
+ AutoMutex lock(mLock);
+ sp<VolumeShaper::State> state = mAudioTrack->getVolumeShaperState(id);
+ if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
+ if (restoreTrack_l("getVolumeShaperState") == OK) {
+ state = mAudioTrack->getVolumeShaperState(id);
+ }
+ }
+ return state;
+}
+
+status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
+{
+ if (timestamp == nullptr) {
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ return getTimestamp_l(timestamp);
+}
+
+status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
+{
+ if (mCblk->mFlags & CBLK_INVALID) {
+ const status_t status = restoreTrack_l("getTimestampExtended");
+ if (status != OK) {
+ // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
+ // recommending that the track be recreated.
+ return DEAD_OBJECT;
+ }
+ }
+ // check for offloaded/direct here in case restoring somehow changed those flags.
+ if (isOffloadedOrDirect_l()) {
+ return INVALID_OPERATION; // not supported
+ }
+ status_t status = mProxy->getTimestamp(timestamp);
+ LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
+ __func__, mPortId, status);
+ bool found = false;
+ timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
+ timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
+ // server side frame offset in case AudioTrack has been restored.
+ for (int i = ExtendedTimestamp::LOCATION_SERVER;
+ i < ExtendedTimestamp::LOCATION_MAX; ++i) {
+ if (timestamp->mTimeNs[i] >= 0) {
+ // apply server offset (frames flushed is ignored
+ // so we don't report the jump when the flush occurs).
+ timestamp->mPosition[i] += mFramesWrittenServerOffset;
+ found = true;
+ }
+ }
+ return found ? OK : WOULD_BLOCK;
+}
+
+status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
+{
+ AutoMutex lock(mLock);
+ return getTimestamp_l(timestamp);
+}
+
+status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
+{
+ bool previousTimestampValid = mPreviousTimestampValid;
+ // Set false here to cover all the error return cases.
+ mPreviousTimestampValid = false;
+
+ switch (mState) {
+ case STATE_ACTIVE:
+ case STATE_PAUSED:
+ break; // handle below
+ case STATE_FLUSHED:
+ case STATE_STOPPED:
+ return WOULD_BLOCK;
+ case STATE_STOPPING:
+ case STATE_PAUSED_STOPPING:
+ if (!isOffloaded_l()) {
+ return INVALID_OPERATION;
+ }
+ break; // offloaded tracks handled below
+ default:
+ LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
+ __func__, mPortId, mState);
+ break;
+ }
+
+ if (mCblk->mFlags & CBLK_INVALID) {
+ const status_t status = restoreTrack_l("getTimestamp");
+ if (status != OK) {
+ // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
+ // recommending that the track be recreated.
+ return DEAD_OBJECT;
+ }
+ }
+
+ // The presented frame count must always lag behind the consumed frame count.
+ // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
+
+ status_t status;
+ if (isOffloadedOrDirect_l()) {
+ // use Binder to get timestamp
+ status = mAudioTrack->getTimestamp(timestamp);
+ } else {
+ // read timestamp from shared memory
+ ExtendedTimestamp ets;
+ status = mProxy->getTimestamp(&ets);
+ if (status == OK) {
+ ExtendedTimestamp::Location location;
+ status = ets.getBestTimestamp(×tamp, &location);
+
+ if (status == OK) {
+ updateLatency_l();
+ // It is possible that the best location has moved from the kernel to the server.
+ // In this case we adjust the position from the previous computed latency.
+ if (location == ExtendedTimestamp::LOCATION_SERVER) {
+ ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
+ "%s(%d): location moved from kernel to server",
+ __func__, mPortId);
+ // check that the last kernel OK time info exists and the positions
+ // are valid (if they predate the current track, the positions may
+ // be zero or negative).
+ const int64_t frames =
+ (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
+ ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
+ ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
+ ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
+ ?
+ int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
+ / 1000)
+ :
+ (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
+ - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
+ ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
+ __func__, mPortId, (long long)frames, ets.toString().c_str());
+ if (frames >= ets.mPosition[location]) {
+ timestamp.mPosition = 0;
+ } else {
+ timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
+ }
+ } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
+ ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
+ "%s(%d): location moved from server to kernel",
+ __func__, mPortId);
+
+ if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
+ ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
+ // In Q, we don't return errors as an invalid time
+ // but instead we leave the last kernel good timestamp alone.
+ //
+ // If server is identical to kernel, the device data pipeline is idle.
+ // A better start time is now. The retrograde check ensures
+ // timestamp monotonicity.
+ const int64_t nowNs = systemTime();
+ if (!mTimestampStallReported) {
+ ALOGD("%s(%d): device stall time corrected using current time %lld",
+ __func__, mPortId, (long long)nowNs);
+ mTimestampStallReported = true;
+ }
+ timestamp.mTime = convertNsToTimespec(nowNs);
+ } else {
+ mTimestampStallReported = false;
+ }
+ }
+
+ // We update the timestamp time even when paused.
+ if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
+ const int64_t now = systemTime();
+ const int64_t at = audio_utils_ns_from_timespec(×tamp.mTime);
+ const int64_t lag =
+ (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
+ ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
+ ? int64_t(mAfLatency * 1000000LL)
+ : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
+ - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
+ * NANOS_PER_SECOND / mSampleRate;
+ const int64_t limit = now - lag; // no earlier than this limit
+ if (at < limit) {
+ ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
+ (long long)lag, (long long)at, (long long)limit);
+ timestamp.mTime = convertNsToTimespec(limit);
+ }
+ }
+ mPreviousLocation = location;
+ } else {
+ // right after AudioTrack is started, one may not find a timestamp
+ ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
+ }
+ }
+ if (status == INVALID_OPERATION) {
+ // INVALID_OPERATION occurs when no timestamp has been issued by the server;
+ // other failures are signaled by a negative time.
+ // If we come out of FLUSHED or STOPPED where the position is known
+ // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
+ // "zero" for NuPlayer). We don't convert for track restoration as position
+ // does not reset.
+ ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
+ __func__, mPortId,
+ (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
+ if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
+ status = WOULD_BLOCK;
+ }
+ }
+ }
+ if (status != NO_ERROR) {
+ ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
+ return status;
+ }
+ if (isOffloadedOrDirect_l()) {
+ if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
+ // use cached paused position in case another offloaded track is running.
+ timestamp.mPosition = mPausedPosition;
+ clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime);
+ // TODO: adjust for delay
+ return NO_ERROR;
+ }
+
+ // Check whether a pending flush or stop has completed, as those commands may
+ // be asynchronous or return near finish or exhibit glitchy behavior.
+ //
+ // Originally this showed up as the first timestamp being a continuation of
+ // the previous song under gapless playback.
+ // However, we sometimes see zero timestamps, then a glitch of
+ // the previous song's position, and then correct timestamps afterwards.
+ if (mStartFromZeroUs != 0 && mSampleRate != 0) {
+ static const int kTimeJitterUs = 100000; // 100 ms
+ static const int k1SecUs = 1000000;
+
+ const int64_t timeNow = getNowUs();
+
+ if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
+ const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
+ if (timestampTimeUs < mStartFromZeroUs) {
+ return WOULD_BLOCK; // stale timestamp time, occurs before start.
+ }
+ const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
+ const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
+ / ((double)mSampleRate * mPlaybackRate.mSpeed);
+
+ if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
+ // Verify that the counter can't count faster than the sample rate
+ // since the start time. If greater, then that means we may have failed
+ // to completely flush or stop the previous playing track.
+ ALOGW_IF(!mTimestampStartupGlitchReported,
+ "%s(%d): startup glitch detected"
+ " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
+ __func__, mPortId,
+ (long long)deltaTimeUs, (long long)deltaPositionByUs,
+ timestamp.mPosition);
+ mTimestampStartupGlitchReported = true;
+ if (previousTimestampValid
+ && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
+ timestamp = mPreviousTimestamp;
+ mPreviousTimestampValid = true;
+ return NO_ERROR;
+ }
+ return WOULD_BLOCK;
+ }
+ if (deltaPositionByUs != 0) {
+ mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
+ }
+ } else {
+ mStartFromZeroUs = 0; // don't check again, start time expired.
+ }
+ mTimestampStartupGlitchReported = false;
+ }
+ } else {
+ // Update the mapping between local consumed (mPosition) and server consumed (mServer)
+ (void) updateAndGetPosition_l();
+ // Server consumed (mServer) and presented both use the same server time base,
+ // and server consumed is always >= presented.
+ // The delta between these represents the number of frames in the buffer pipeline.
+ // If this delta between these is greater than the client position, it means that
+ // actually presented is still stuck at the starting line (figuratively speaking),
+ // waiting for the first frame to go by. So we can't report a valid timestamp yet.
+ // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
+ // mPosition exceeds 32 bits.
+ // TODO Remove when timestamp is updated to contain pipeline status info.
+ const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
+ if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
+ && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
+ return INVALID_OPERATION;
+ }
+ // Convert timestamp position from server time base to client time base.
+ // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
+ // But if we change it to 64-bit then this could fail.
+ // Use Modulo computation here.
+ timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
+ // Immediately after a call to getPosition_l(), mPosition and
+ // mServer both represent the same frame position. mPosition is
+ // in client's point of view, and mServer is in server's point of
+ // view. So the difference between them is the "fudge factor"
+ // between client and server views due to stop() and/or new
+ // IAudioTrack. And timestamp.mPosition is initially in server's
+ // point of view, so we need to apply the same fudge factor to it.
+ }
+
+ // Prevent retrograde motion in timestamp.
+ // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
+ if (status == NO_ERROR) {
+ // Fix stale time when checking timestamp right after start().
+ // The position is at the last reported location but the time can be stale
+ // due to pause or standby or cold start latency.
+ //
+ // We keep advancing the time (but not the position) to ensure that the
+ // stale value does not confuse the application.
+ //
+ // For offload compatibility, use a default lag value here.
+ // Any time discrepancy between this update and the pause timestamp is handled
+ // by the retrograde check afterwards.
+ int64_t currentTimeNanos = audio_utils_ns_from_timespec(×tamp.mTime);
+ const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
+ const int64_t limitNs = mStartNs - lagNs;
+ if (currentTimeNanos < limitNs) {
+ if (!mTimestampStaleTimeReported) {
+ ALOGD("%s(%d): stale timestamp time corrected, "
+ "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
+ __func__, mPortId,
+ (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
+ mTimestampStaleTimeReported = true;
+ }
+ timestamp.mTime = convertNsToTimespec(limitNs);
+ currentTimeNanos = limitNs;
+ } else {
+ mTimestampStaleTimeReported = false;
+ }
+
+ // previousTimestampValid is set to false when starting after a stop or flush.
+ if (previousTimestampValid) {
+ const int64_t previousTimeNanos =
+ audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
+
+ // retrograde check
+ if (currentTimeNanos < previousTimeNanos) {
+ if (!mTimestampRetrogradeTimeReported) {
+ ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
+ __func__, mPortId,
+ (long long)currentTimeNanos, (long long)previousTimeNanos);
+ mTimestampRetrogradeTimeReported = true;
+ }
+ timestamp.mTime = mPreviousTimestamp.mTime;
+ } else {
+ mTimestampRetrogradeTimeReported = false;
+ }
+
+ // Looking at signed delta will work even when the timestamps
+ // are wrapping around.
+ int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
+ - mPreviousTimestamp.mPosition).signedValue();
+ if (deltaPosition < 0) {
+ // Only report once per position instead of spamming the log.
+ if (!mTimestampRetrogradePositionReported) {
+ ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
+ __func__, mPortId,
+ deltaPosition,
+ timestamp.mPosition,
+ mPreviousTimestamp.mPosition);
+ mTimestampRetrogradePositionReported = true;
+ }
+ } else {
+ mTimestampRetrogradePositionReported = false;
+ }
+ if (deltaPosition < 0) {
+ timestamp.mPosition = mPreviousTimestamp.mPosition;
+ deltaPosition = 0;
+ }
+#if 0
+ // Uncomment this to verify audio timestamp rate.
+ const int64_t deltaTime =
+ audio_utils_ns_from_timespec(×tamp.mTime) - previousTimeNanos;
+ if (deltaTime != 0) {
+ const int64_t computedSampleRate =
+ deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
+ ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
+ __func__, mPortId,
+ (unsigned)computedSampleRate, mSampleRate);
+ }
+#endif
+ }
+ mPreviousTimestamp = timestamp;
+ mPreviousTimestampValid = true;
+ }
+
+ return status;
+}
+
+String8 AudioTrack::getParameters(const String8& keys)
+{
+ audio_io_handle_t output = getOutput();
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ return AudioSystem::getParameters(output, keys);
+ } else {
+ return String8::empty();
+ }
+}
+
+bool AudioTrack::isOffloaded() const
+{
+ AutoMutex lock(mLock);
+ return isOffloaded_l();
+}
+
+bool AudioTrack::isDirect() const
+{
+ AutoMutex lock(mLock);
+ return isDirect_l();
+}
+
+bool AudioTrack::isOffloadedOrDirect() const
+{
+ AutoMutex lock(mLock);
+ return isOffloadedOrDirect_l();
+}
+
+
+status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
+{
+ String8 result;
+
+ result.append(" AudioTrack::dump\n");
+ result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
+ mPortId, mStatus, mState, mSessionId, mFlags);
+ result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
+ (mStreamType == AUDIO_STREAM_DEFAULT) ?
+ AudioSystem::attributesToStreamType(mAttributes) :
+ mStreamType,
+ mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
+ result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
+ mFormat, mChannelMask, mChannelCount);
+ result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
+ mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
+ result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
+ mFrameCount, mReqFrameCount);
+ result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
+ " req. notif. per buff(%u)\n",
+ mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
+ result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
+ mLatency, mSelectedDeviceId, mRoutedDeviceId);
+ result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
+ mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
+ ::write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+uint32_t AudioTrack::getUnderrunCount() const
+{
+ AutoMutex lock(mLock);
+ return getUnderrunCount_l();
+}
+
+uint32_t AudioTrack::getUnderrunCount_l() const
+{
+ return mProxy->getUnderrunCount() + mUnderrunCountOffset;
+}
+
+uint32_t AudioTrack::getUnderrunFrames() const
+{
+ AutoMutex lock(mLock);
+ return mProxy->getUnderrunFrames();
+}
+
+status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
+{
+
+ if (callback == 0) {
+ ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ if (mDeviceCallback.unsafe_get() == callback.get()) {
+ ALOGW("%s(%d): adding same callback!", __func__, mPortId);
+ return INVALID_OPERATION;
+ }
+ status_t status = NO_ERROR;
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ if (mDeviceCallback != 0) {
+ ALOGW("%s(%d): callback already present!", __func__, mPortId);
+ AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
+ }
+ status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
+ }
+ mDeviceCallback = callback;
+ return status;
+}
+
+status_t AudioTrack::removeAudioDeviceCallback(
+ const sp<AudioSystem::AudioDeviceCallback>& callback)
+{
+ if (callback == 0) {
+ ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ if (mDeviceCallback.unsafe_get() != callback.get()) {
+ ALOGW("%s removing different callback!", __FUNCTION__);
+ return INVALID_OPERATION;
+ }
+ mDeviceCallback.clear();
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
+ }
+ return NO_ERROR;
+}
+
+
+void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
+ audio_port_handle_t deviceId)
+{
+ sp<AudioSystem::AudioDeviceCallback> callback;
+ {
+ AutoMutex lock(mLock);
+ if (audioIo != mOutput) {
+ return;
+ }
+ callback = mDeviceCallback.promote();
+ // only update device if the track is active as route changes due to other use cases are
+ // irrelevant for this client
+ if (mState == STATE_ACTIVE) {
+ mRoutedDeviceId = deviceId;
+ }
+ }
+
+ if (callback.get() != nullptr) {
+ callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
+ }
+}
+
+status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
+{
+ if (msec == nullptr ||
+ (location != ExtendedTimestamp::LOCATION_SERVER
+ && location != ExtendedTimestamp::LOCATION_KERNEL)) {
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ // inclusive of offloaded and direct tracks.
+ //
+ // It is possible, but not enabled, to allow duration computation for non-pcm
+ // audio_has_proportional_frames() formats because currently they have
+ // the drain rate equivalent to the pcm sample rate * framesize.
+ if (!isPurePcmData_l()) {
+ return INVALID_OPERATION;
+ }
+ ExtendedTimestamp ets;
+ if (getTimestamp_l(&ets) == OK
+ && ets.mTimeNs[location] > 0) {
+ int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
+ - ets.mPosition[location];
+ if (diff < 0) {
+ *msec = 0;
+ } else {
+ // ms is the playback time by frames
+ int64_t ms = (int64_t)((double)diff * 1000 /
+ ((double)mSampleRate * mPlaybackRate.mSpeed));
+ // clockdiff is the timestamp age (negative)
+ int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
+ ets.mTimeNs[location]
+ + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
+ - systemTime(SYSTEM_TIME_MONOTONIC);
+
+ //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
+ static const int NANOS_PER_MILLIS = 1000000;
+ *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
+ }
+ return NO_ERROR;
+ }
+ if (location != ExtendedTimestamp::LOCATION_SERVER) {
+ return INVALID_OPERATION; // LOCATION_KERNEL is not available
+ }
+ // use server position directly (offloaded and direct arrive here)
+ updateAndGetPosition_l();
+ int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
+ *msec = (diff <= 0) ? 0
+ : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
+ return NO_ERROR;
+}
+
+bool AudioTrack::hasStarted()
+{
+ AutoMutex lock(mLock);
+ switch (mState) {
+ case STATE_STOPPED:
+ if (isOffloadedOrDirect_l()) {
+ // check if we have started in the past to return true.
+ return mStartFromZeroUs > 0;
+ }
+ // A normal audio track may still be draining, so
+ // check if stream has ended. This covers fasttrack position
+ // instability and start/stop without any data written.
+ if (mProxy->getStreamEndDone()) {
+ return true;
+ }
+ FALLTHROUGH_INTENDED;
+ case STATE_ACTIVE:
+ case STATE_STOPPING:
+ break;
+ case STATE_PAUSED:
+ case STATE_PAUSED_STOPPING:
+ case STATE_FLUSHED:
+ return false; // we're not active
+ default:
+ LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
+ break;
+ }
+
+ // wait indicates whether we need to wait for a timestamp.
+ // This is conservatively figured - if we encounter an unexpected error
+ // then we will not wait.
+ bool wait = false;
+ if (isOffloadedOrDirect_l()) {
+ AudioTimestamp ts;
+ status_t status = getTimestamp_l(ts);
+ if (status == WOULD_BLOCK) {
+ wait = true;
+ } else if (status == OK) {
+ wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
+ }
+ ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
+ __func__, mPortId,
+ (int)wait,
+ ts.mPosition,
+ (long long)mStartTs.mPosition);
+ } else {
+ int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
+ ExtendedTimestamp ets;
+ status_t status = getTimestamp_l(&ets);
+ if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
+ wait = true;
+ } else if (status == OK) {
+ for (location = ExtendedTimestamp::LOCATION_KERNEL;
+ location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
+ if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
+ continue;
+ }
+ wait = ets.mPosition[location] == 0
+ || ets.mPosition[location] == mStartEts.mPosition[location];
+ break;
+ }
+ }
+ ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
+ __func__, mPortId,
+ (int)wait,
+ (long long)ets.mPosition[location],
+ (long long)mStartEts.mPosition[location]);
+ }
+ return !wait;
+}
+
+// =========================================================================
+
+void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
+{
+ sp<AudioTrack> audioTrack = mAudioTrack.promote();
+ if (audioTrack != 0) {
+ AutoMutex lock(audioTrack->mLock);
+ audioTrack->mProxy->binderDied();
+ }
+}
+
+// =========================================================================
+
+AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
+ : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
+ , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
+ mIgnoreNextPausedInt(false)
+{
+}
+
+AudioTrack::AudioTrackThread::~AudioTrackThread()
+{
+}
+
+bool AudioTrack::AudioTrackThread::threadLoop()
+{
+ {
+ AutoMutex _l(mMyLock);
+ if (mPaused) {
+ // TODO check return value and handle or log
+ mMyCond.wait(mMyLock);
+ // caller will check for exitPending()
+ return true;
+ }
+ if (mIgnoreNextPausedInt) {
+ mIgnoreNextPausedInt = false;
+ mPausedInt = false;
+ }
+ if (mPausedInt) {
+ // TODO use futex instead of condition, for event flag "or"
+ if (mPausedNs > 0) {
+ // TODO check return value and handle or log
+ (void) mMyCond.waitRelative(mMyLock, mPausedNs);
+ } else {
+ // TODO check return value and handle or log
+ mMyCond.wait(mMyLock);
+ }
+ mPausedInt = false;
+ return true;
+ }
+ }
+ if (exitPending()) {
+ return false;
+ }
+ nsecs_t ns = mReceiver.processAudioBuffer();
+ switch (ns) {
+ case 0:
+ return true;
+ case NS_INACTIVE:
+ pauseInternal();
+ return true;
+ case NS_NEVER:
+ return false;
+ case NS_WHENEVER:
+ // Event driven: call wake() when callback notifications conditions change.
+ ns = INT64_MAX;
+ FALLTHROUGH_INTENDED;
+ default:
+ LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
+ __func__, mReceiver.mPortId, (long long)ns);
+ pauseInternal(ns);
+ return true;
+ }
+}
+
+void AudioTrack::AudioTrackThread::requestExit()
+{
+ // must be in this order to avoid a race condition
+ Thread::requestExit();
+ resume();
+}
+
+void AudioTrack::AudioTrackThread::pause()
+{
+ AutoMutex _l(mMyLock);
+ mPaused = true;
+}
+
+void AudioTrack::AudioTrackThread::resume()
+{
+ AutoMutex _l(mMyLock);
+ mIgnoreNextPausedInt = true;
+ if (mPaused || mPausedInt) {
+ mPaused = false;
+ mPausedInt = false;
+ mMyCond.signal();
+ }
+}
+
+void AudioTrack::AudioTrackThread::wake()
+{
+ AutoMutex _l(mMyLock);
+ if (!mPaused) {
+ // wake() might be called while servicing a callback - ignore the next
+ // pause time and call processAudioBuffer.
+ mIgnoreNextPausedInt = true;
+ if (mPausedInt && mPausedNs > 0) {
+ // audio track is active and internally paused with timeout.
+ mPausedInt = false;
+ mMyCond.signal();
+ }
+ }
+}
+
+void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
+{
+ AutoMutex _l(mMyLock);
+ mPausedInt = true;
+ mPausedNs = ns;
+}
+
+} // namespace android
diff --git a/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/poc.cpp b/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/poc.cpp
new file mode 100644
index 0000000..0ecd918
--- /dev/null
+++ b/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/poc.cpp
@@ -0,0 +1,126 @@
+/**
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <media/AudioSystem.h>
+#include <media/AudioTrack.h>
+#include <binder/MemoryDealer.h>
+#include <binder/ProcessState.h>
+#include <math.h>
+#include "poc.h"
+
+namespace android {
+
+AudioTrackTest::AudioTrackTest(void) {
+ InitSine();
+}
+
+#define BUF_SZ 999999
+#define TIMEOUT_SEC 3 * 60
+
+int AudioTrackTest::Test01() {
+ sp < MemoryDealer > heap;
+ sp < IMemory > iMem;
+ audio_track_cblk_t* p;
+
+ unsigned long smpBuf[BUF_SZ];
+ unsigned long rate = 48000;
+ unsigned long phi;
+ unsigned long dPhi;
+ unsigned long amplitude;
+ unsigned long freq = 1237;
+ unsigned f0;
+
+ f0 = pow(2., 32.) * freq / rate;
+ dPhi = (unsigned long) f0;
+ amplitude = 1000;
+ phi = 0;
+ Generate(smpBuf, BUF_SZ, amplitude, phi, dPhi);
+
+ heap = new MemoryDealer(7999992, "AudioTrack Heap Base");
+ iMem = heap->allocate(BUF_SZ * sizeof(unsigned long));
+
+ p = static_cast<audio_track_cblk_t*>(iMem->pointer());
+ memcpy(p, smpBuf, BUF_SZ * sizeof(unsigned long));
+
+ sp < AudioTrack > track = new AudioTrack(AUDIO_STREAM_MUSIC,
+ rate, AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_OUT_STEREO, iMem);
+
+ status_t status = track->initCheck();
+ if (status != NO_ERROR) {
+ track.clear();
+ return EXIT_FAILURE;
+ }
+ track->start();
+ sleep(TIMEOUT_SEC);
+ track->stop();
+ iMem.clear();
+ heap.clear();
+ return EXIT_SUCCESS;
+}
+
+void AudioTrackTest::Generate(unsigned long *buffer, unsigned long bufferSz,
+ unsigned long amplitude, unsigned long &phi,
+ unsigned long dPhi) {
+ for (unsigned long i0 = 0; i0 < bufferSz; i0++) {
+ buffer[i0] = ComputeSine(amplitude, phi);
+ phi += dPhi;
+ }
+}
+
+unsigned long AudioTrackTest::ComputeSine(unsigned long amplitude,
+ unsigned long phi) {
+ unsigned long pi13 = 25736;
+ unsigned long sample;
+ unsigned long l0, l1;
+
+ sample = (amplitude * sin1024[(phi >> 22) & 0x3ff]) >> 15;
+ l0 = (phi >> 12) & 0x3ff;
+ l1 = (amplitude * sin1024[((phi >> 22) + 256) & 0x3ff]) >> 15;
+ l0 = (l0 * l1) >> 10;
+ l0 = (l0 * pi13) >> 22;
+ sample = sample + l0;
+
+ return (unsigned long) sample;
+}
+
+void AudioTrackTest::InitSine(void) {
+ unsigned phi = 0;
+ unsigned dPhi = 2 * M_PI / SIN_SZ;
+ for (unsigned i0 = 0; i0 < SIN_SZ; i0++) {
+ long d0;
+
+ d0 = 32768. * sin(phi);
+ phi += dPhi;
+ if (d0 >= 32767)
+ d0 = 32767;
+ if (d0 <= -32768)
+ d0 = -32768;
+ sin1024[i0] = (long) d0;
+ }
+}
+}
+
+using namespace android;
+int main() {
+ ProcessState::self()->startThreadPool();
+ AudioTrackTest *test;
+
+ test = new AudioTrackTest();
+ test->Test01();
+ delete test;
+
+ return EXIT_SUCCESS;
+}
diff --git a/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/poc.h b/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/poc.h
new file mode 100644
index 0000000..ecaf48f
--- /dev/null
+++ b/hostsidetests/securitybulletin/securityPatch/CVE-2017-0597/poc.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AUDIOTRACKTEST_H_
+#define AUDIOTRACKTEST_H_
+
+namespace android {
+
+class AudioTrackTest {
+public:
+ AudioTrackTest(void);
+ ~AudioTrackTest(){};
+
+ void Execute(void);
+ int Test01();
+
+ void Generate(unsigned long *buffer, unsigned long bufferSz,
+ unsigned long amplitude, unsigned long &phi,
+ unsigned long dPhi);
+ void InitSine();
+ unsigned long ComputeSine(unsigned long amplitude, unsigned long phi);
+
+#define SIN_SZ 5200000
+ unsigned long sin1024[SIN_SZ];
+};
+}; // namespace android
+
+#endif /*AUDIOTRACKTEST_H_*/
diff --git a/hostsidetests/securitybulletin/src/android/security/cts/CVE_2017_0597.java b/hostsidetests/securitybulletin/src/android/security/cts/CVE_2017_0597.java
new file mode 100644
index 0000000..7f0bd0d
--- /dev/null
+++ b/hostsidetests/securitybulletin/src/android/security/cts/CVE_2017_0597.java
@@ -0,0 +1,37 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.security.cts;
+import android.platform.test.annotations.SecurityTest;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import com.android.tradefed.testtype.DeviceJUnit4ClassRunner;
+
+@RunWith(DeviceJUnit4ClassRunner.class)
+public class CVE_2017_0597 extends SecurityTestCase {
+
+ /**
+ * b/34749571
+ * Vulnerability Behaviour: SIGSEGV in audioserver
+ **/
+ @Test
+ @SecurityTest(minPatchLevel = "2017-05")
+ public void testPocCVE_2017_0597() throws Exception {
+ String processPatternStrings[] = {"audioserver"};
+ AdbUtils.runPocAssertNoCrashesNotVulnerable("CVE-2017-0597", null, getDevice(),
+ processPatternStrings);
+ }
+}
