AudioTrack.cpp revision 3bcffa136909c1fb6e88ee4efd12ccac18360a85
198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams/* 298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** 398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** Copyright 2007, The Android Open Source Project 498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** 598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** Licensed under the Apache License, Version 2.0 (the "License"); 698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** you may not use this file except in compliance with the License. 798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** You may obtain a copy of the License at 898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** 998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** http://www.apache.org/licenses/LICENSE-2.0 1098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** 1198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** Unless required by applicable law or agreed to in writing, software 1298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** distributed under the License is distributed on an "AS IS" BASIS, 1398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 1498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** See the License for the specific language governing permissions and 1598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams** limitations under the License. 1698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams*/ 1798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 1898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 190f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines//#define LOG_NDEBUG 0 2098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams#define LOG_TAG "AudioTrack" 2198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 2298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams#include <math.h> 237d435ae5ba100be5710b685653cc351cab159c11Stephen Hines#include <sys/resource.h> 247d435ae5ba100be5710b685653cc351cab159c11Stephen Hines#include <audio_utils/primitives.h> 257d435ae5ba100be5710b685653cc351cab159c11Stephen Hines#include <binder/IPCThreadState.h> 267d435ae5ba100be5710b685653cc351cab159c11Stephen Hines#include <media/AudioTrack.h> 277d435ae5ba100be5710b685653cc351cab159c11Stephen Hines#include <utils/Log.h> 2898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams#include <private/media/AudioTrackShared.h> 2998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams#include <media/IAudioFlinger.h> 3098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 31eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray#define WAIT_PERIOD_MS 10 32ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray#define WAIT_STREAM_END_TIMEOUT_SEC 120 3398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 3498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 3598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Samsnamespace android { 3698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams// --------------------------------------------------------------------------- 370f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines 38eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray// static 39ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murraystatus_t AudioTrack::getMinFrameCount( 4098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams size_t* frameCount, 4198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_stream_type_t streamType, 4298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams uint32_t sampleRate) 430f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines{ 44ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (frameCount == NULL) { 45ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray return BAD_VALUE; 46ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 4798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 4898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // FIXME merge with similar code in createTrack_l(), except we're missing 4998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // some information here that is available in createTrack_l(): 5098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // audio_io_handle_t output 5198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // audio_format_t format 5298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // audio_channel_mask_t channelMask 5398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // audio_output_flags_t flags 5498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams uint32_t afSampleRate; 5598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams status_t status; 5698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 570f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines if (status != NO_ERROR) { 58ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray ALOGE("Unable to query output sample rate for stream type %d; status %d", 59ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray streamType, status); 60ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray return status; 6198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 6298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams size_t afFrameCount; 6398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 640f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines if (status != NO_ERROR) { 65ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray ALOGE("Unable to query output frame count for stream type %d; status %d", 66ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray streamType, status); 67ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray return status; 6898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 6998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams uint32_t afLatency; 7098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams status = AudioSystem::getOutputLatency(&afLatency, streamType); 7198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (status != NO_ERROR) { 7298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("Unable to query output latency for stream type %d; status %d", 7398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams streamType, status); 7498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return status; 750f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines } 76ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 77ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray // Ensure that buffer depth covers at least audio hardware latency 78ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 7998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (minBufCount < 2) { 8098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams minBufCount = 2; 8198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 8298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 8398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 8498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams afFrameCount * minBufCount * sampleRate / afSampleRate; 8598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // The formula above should always produce a non-zero value, but return an error 8698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // in the unlikely event that it does not, as that's part of the API contract. 870f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines if (*frameCount == 0) { 88ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 89ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray streamType, sampleRate); 90ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray return BAD_VALUE; 9198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 9298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 9398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 9498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return NO_ERROR; 950f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines} 96ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 97ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray// --------------------------------------------------------------------------- 98ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 9998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason SamsAudioTrack::AudioTrack() 10098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams : mStatus(NO_INIT), 10198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mIsTimed(false), 10298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPreviousPriority(ANDROID_PRIORITY_NORMAL), 10398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPreviousSchedulingGroup(SP_DEFAULT), 10498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPausedPosition(0) 1050f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines{ 106ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray} 107ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 108ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim MurrayAudioTrack::AudioTrack( 10998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_stream_type_t streamType, 11098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams uint32_t sampleRate, 11198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_format_t format, 11298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_channel_mask_t channelMask, 11398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams size_t frameCount, 11498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_output_flags_t flags, 11598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams callback_t cbf, 11698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams void* user, 1170f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines uint32_t notificationFrames, 118ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray int sessionId, 119ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray transfer_type transferType, 120ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray const audio_offload_info_t *offloadInfo, 12198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams int uid, 12298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams pid_t pid) 12398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams : mStatus(NO_INIT), 12498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mIsTimed(false), 12598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPreviousPriority(ANDROID_PRIORITY_NORMAL), 12698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPreviousSchedulingGroup(SP_DEFAULT), 12798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPausedPosition(0) 12898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams{ 12998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mStatus = set(streamType, sampleRate, format, channelMask, 13098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams frameCount, flags, cbf, user, notificationFrames, 13198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 13298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams offloadInfo, uid, pid); 1330f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines} 134ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 135ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim MurrayAudioTrack::AudioTrack( 136ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray audio_stream_type_t streamType, 13798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams uint32_t sampleRate, 13898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_format_t format, 13998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_channel_mask_t channelMask, 14098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams const sp<IMemory>& sharedBuffer, 14198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_output_flags_t flags, 14298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams callback_t cbf, 14398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams void* user, 1440f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines uint32_t notificationFrames, 145ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray int sessionId, 146ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray transfer_type transferType, 147ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray const audio_offload_info_t *offloadInfo, 14898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams int uid, 14998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams pid_t pid) 15098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams : mStatus(NO_INIT), 15198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mIsTimed(false), 15298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPreviousPriority(ANDROID_PRIORITY_NORMAL), 15398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPreviousSchedulingGroup(SP_DEFAULT), 15498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPausedPosition(0) 15598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams{ 15698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mStatus = set(streamType, sampleRate, format, channelMask, 15798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 0 /*frameCount*/, flags, cbf, user, notificationFrames, 1580f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 159ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray uid, pid); 160ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray} 161ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 16298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason SamsAudioTrack::~AudioTrack() 16398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams{ 16498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (mStatus == NO_ERROR) { 16598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // Make sure that callback function exits in the case where 16698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // it is looping on buffer full condition in obtainBuffer(). 16798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // Otherwise the callback thread will never exit. 16898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams stop(); 16998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (mAudioTrackThread != 0) { 17098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mProxy->interrupt(); 17198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 17298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mAudioTrackThread->requestExitAndWait(); 17398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mAudioTrackThread.clear(); 1740f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines } 175ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 176ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mAudioTrack.clear(); 177ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mCblkMemory.clear(); 17898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mSharedBuffer.clear(); 17998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams IPCThreadState::self()->flushCommands(); 18098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 18198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams IPCThreadState::self()->getCallingPid(), mClientPid); 18298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 18398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 18498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams} 18598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 18698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Samsstatus_t AudioTrack::set( 18798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_stream_type_t streamType, 18898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams uint32_t sampleRate, 18998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_format_t format, 19098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_channel_mask_t channelMask, 19198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams size_t frameCount, 19298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams audio_output_flags_t flags, 19398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams callback_t cbf, 1940f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines void* user, 195ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray uint32_t notificationFrames, 196ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray const sp<IMemory>& sharedBuffer, 197ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray bool threadCanCallJava, 19898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams int sessionId, 19998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams transfer_type transferType, 20098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams const audio_offload_info_t *offloadInfo, 20198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams int uid, 2020f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines pid_t pid) 203ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray{ 204ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 205ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 20698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 20798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams sessionId, transferType); 20898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 20998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams switch (transferType) { 2100f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines case TRANSFER_DEFAULT: 211eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray if (sharedBuffer != 0) { 212ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray transferType = TRANSFER_SHARED; 213ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } else if (cbf == NULL || threadCanCallJava) { 21498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams transferType = TRANSFER_SYNC; 21598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } else { 21698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams transferType = TRANSFER_CALLBACK; 21798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 21898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams break; 21998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams case TRANSFER_CALLBACK: 22098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (cbf == NULL || sharedBuffer != 0) { 22198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 2220f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines return BAD_VALUE; 223eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } 224ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray break; 225ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray case TRANSFER_OBTAIN: 22698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams case TRANSFER_SYNC: 22798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (sharedBuffer != 0) { 22898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 22998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return BAD_VALUE; 2300f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines } 231eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray break; 232ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray case TRANSFER_SHARED: 233ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (sharedBuffer == 0) { 23498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 23598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return BAD_VALUE; 23698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 23798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams break; 23898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams default: 2390f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines ALOGE("Invalid transfer type %d", transferType); 240eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray return BAD_VALUE; 241ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 242ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mSharedBuffer = sharedBuffer; 24398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mTransfer = transferType; 24498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 24598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 24698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams sharedBuffer->size()); 24798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 24898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 24998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 2500f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines AutoMutex lock(mLock); 251eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray 252ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray // invariant that mAudioTrack != 0 is true only after set() returns successfully 253ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (mAudioTrack != 0) { 25498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("Track already in use"); 25598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return INVALID_OPERATION; 25698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 25798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 2580f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines // handle default values first. 259eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray if (streamType == AUDIO_STREAM_DEFAULT) { 260ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray streamType = AUDIO_STREAM_MUSIC; 261ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 26298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 26398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("Invalid stream type %d", streamType); 26498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return BAD_VALUE; 26598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 26698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mStreamType = streamType; 2670f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines 268eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray status_t status; 269ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (sampleRate == 0) { 270ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); 27198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (status != NO_ERROR) { 27298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("Could not get output sample rate for stream type %d; status %d", 27398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams streamType, status); 27498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return status; 27598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 27698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 27798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mSampleRate = sampleRate; 2780f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines 279eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray // these below should probably come from the audioFlinger too... 280ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (format == AUDIO_FORMAT_DEFAULT) { 281ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray format = AUDIO_FORMAT_PCM_16_BIT; 28298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 28398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 28498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // validate parameters 28598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (!audio_is_valid_format(format)) { 2860f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines ALOGE("Invalid format %#x", format); 287eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray return BAD_VALUE; 288ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 289ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mFormat = format; 29098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 29198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (!audio_is_output_channel(channelMask)) { 29298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("Invalid channel mask %#x", channelMask); 29398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return BAD_VALUE; 29498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 2950f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mChannelMask = channelMask; 296eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 297ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mChannelCount = channelCount; 298ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 29998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // AudioFlinger does not currently support 8-bit data in shared memory 30098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 30198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOGE("8-bit data in shared memory is not supported"); 30298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return BAD_VALUE; 30398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 30498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 30598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // force direct flag if format is not linear PCM 3060f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines // or offload was requested 307eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 308ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray || !audio_is_linear_pcm(format)) { 309ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 31098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ? "Offload request, forcing to Direct Output" 31198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams : "Not linear PCM, forcing to Direct Output"); 31298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams flags = (audio_output_flags_t) 31398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // FIXME why can't we allow direct AND fast? 3140f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 315eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } 316ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray // only allow deep buffering for music stream type 317ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (streamType != AUDIO_STREAM_MUSIC) { 31898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 31998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 32098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 32198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 32298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (audio_is_linear_pcm(format)) { 3230f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mFrameSize = channelCount * audio_bytes_per_sample(format); 324eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } else { 325ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mFrameSize = sizeof(uint8_t); 326ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 32798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mFrameSizeAF = mFrameSize; 32898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } else { 32998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOG_ASSERT(audio_is_linear_pcm(format)); 33098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mFrameSize = channelCount * audio_bytes_per_sample(format); 33198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mFrameSizeAF = channelCount * audio_bytes_per_sample( 33298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 33398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // createTrack will return an error if PCM format is not supported by server, 3340f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines // so no need to check for specific PCM formats here 335eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } 336ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 337ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray // Make copy of input parameter offloadInfo so that in the future: 33898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // (a) createTrack_l doesn't need it as an input parameter 33998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // (b) we can support re-creation of offloaded tracks 34098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (offloadInfo != NULL) { 34198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mOffloadInfoCopy = *offloadInfo; 3420f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mOffloadInfo = &mOffloadInfoCopy; 343eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } else { 344ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mOffloadInfo = NULL; 345ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 34698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 34798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 34898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 34998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mSendLevel = 0.0f; 35098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // mFrameCount is initialized in createTrack_l 3510f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mReqFrameCount = frameCount; 352eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray mNotificationFramesReq = notificationFrames; 353ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mNotificationFramesAct = 0; 354ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mSessionId = sessionId; 35598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams int callingpid = IPCThreadState::self()->getCallingPid(); 35698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams int mypid = getpid(); 35798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (uid == -1 || (callingpid != mypid)) { 35898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mClientUid = IPCThreadState::self()->getCallingUid(); 35998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } else { 36098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mClientUid = uid; 36198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 3620f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines if (pid == -1 || (callingpid != mypid)) { 363eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray mClientPid = callingpid; 364ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } else { 365ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mClientPid = pid; 36698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 36798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mAuxEffectId = 0; 36898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mFlags = flags; 36998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mCbf = cbf; 3700f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines 371eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray if (cbf != NULL) { 372ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 373ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 37498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 37598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 37698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // create the IAudioTrack 37798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams status = createTrack_l(0 /*epoch*/); 37898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 3790f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines if (status != NO_ERROR) { 380eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray if (mAudioTrackThread != 0) { 381ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 382ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mAudioTrackThread->requestExitAndWait(); 38398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mAudioTrackThread.clear(); 38498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 38598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return status; 38698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 38798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 38898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mStatus = NO_ERROR; 38998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mState = STATE_STOPPED; 3900f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mUserData = user; 391eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray mLoopPeriod = 0; 392ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mMarkerPosition = 0; 393ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mMarkerReached = false; 39498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mNewPosition = 0; 39598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mUpdatePeriod = 0; 39698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 39798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mSequence = 1; 3980f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mObservedSequence = mSequence; 399eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray mInUnderrun = false; 400ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 401ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray return NO_ERROR; 40298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams} 40398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 40498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams// ------------------------------------------------------------------------- 40598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 40698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Samsstatus_t AudioTrack::start() 4070f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines{ 408eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray AutoMutex lock(mLock); 409ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 410ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (mState == STATE_ACTIVE) { 41198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return INVALID_OPERATION; 41298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 41398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 41498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mInUnderrun = true; 41598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 41698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams State previousState = mState; 41798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (previousState == STATE_PAUSED_STOPPING) { 4180f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mState = STATE_STOPPING; 419eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } else { 420ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mState = STATE_ACTIVE; 421ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 42298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 42398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // reset current position as seen by client to 0 42498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 42598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // force refresh of remaining frames by processAudioBuffer() as last 4260f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines // write before stop could be partial. 427eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray mRefreshRemaining = true; 428ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 429ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mNewPosition = mProxy->getPosition() + mUpdatePeriod; 43098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 43198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 43298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams sp<AudioTrackThread> t = mAudioTrackThread; 43398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (t != 0) { 43498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (previousState == STATE_STOPPING) { 4350f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mProxy->interrupt(); 436eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } else { 437ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray t->resume(); 438ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 43998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } else { 44098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mPreviousPriority = getpriority(PRIO_PROCESS, 0); 44198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams get_sched_policy(0, &mPreviousSchedulingGroup); 44298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 44398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 44498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 44598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams status_t status = NO_ERROR; 4460f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines if (!(flags & CBLK_INVALID)) { 447eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray status = mAudioTrack->start(); 448ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (status == DEAD_OBJECT) { 449ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray flags |= CBLK_INVALID; 45098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 45198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 45298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (flags & CBLK_INVALID) { 45398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams status = restoreTrack_l("start"); 4540f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines } 455eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray 456ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (status != NO_ERROR) { 457ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray ALOGE("start() status %d", status); 45898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mState = previousState; 45998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (t != 0) { 46098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (previousState != STATE_STOPPING) { 46198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams t->pause(); 46298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 4630f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines } else { 464eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray setpriority(PRIO_PROCESS, 0, mPreviousPriority); 465ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray set_sched_policy(0, mPreviousSchedulingGroup); 466ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 46798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 46898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 46998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return status; 47098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams} 47198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 47298a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Samsvoid AudioTrack::stop() 47398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams{ 4740f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines AutoMutex lock(mLock); 475eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 476ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray return; 477ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 47898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 47998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (isOffloaded_l()) { 48098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mState = STATE_STOPPING; 48198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } else { 4820f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mState = STATE_STOPPED; 483eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } 484ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 485ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mProxy->interrupt(); 48698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mAudioTrack->stop(); 48798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // the playback head position will reset to 0, so if a marker is set, we need 48898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams // to activate it again 48998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mMarkerReached = false; 49098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams#if 0 4910f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines // Force flush if a shared buffer is used otherwise audioflinger 492eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray // will not stop before end of buffer is reached. 493ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray // It may be needed to make sure that we stop playback, likely in case looping is on. 494ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray if (mSharedBuffer != 0) { 49598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams flush_l(); 49698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 49798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams#endif 49898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 49998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams sp<AudioTrackThread> t = mAudioTrackThread; 50098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (t != 0) { 50198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (!isOffloaded_l()) { 5020f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines t->pause(); 503eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray } 504ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } else { 505ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray setpriority(PRIO_PROCESS, 0, mPreviousPriority); 50698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams set_sched_policy(0, mPreviousSchedulingGroup); 50798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 50898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams} 50998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 51098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Samsbool AudioTrack::stopped() const 51198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams{ 5120f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines AutoMutex lock(mLock); 513eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray return mState != STATE_ACTIVE; 514ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray} 515ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray 51698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Samsvoid AudioTrack::flush() 51798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams{ 51898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (mSharedBuffer != 0) { 51998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return; 52098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 52198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams AutoMutex lock(mLock); 5220f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 523eeb3042428fbe3a3cace554d3aca43b324904ad1Tim Murray return; 524ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 525ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray flush_l(); 52698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams} 52798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 52898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Samsvoid AudioTrack::flush_l() 52998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams{ 53098a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams ALOG_ASSERT(mState != STATE_ACTIVE); 53198a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 5320f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines // clear playback marker and periodic update counter 533ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mMarkerPosition = 0; 534ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mMarkerReached = false; 535ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mUpdatePeriod = 0; 53698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mRefreshRemaining = true; 53798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 53898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mState = STATE_FLUSHED; 53998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (isOffloaded_l()) { 5400f5bae87e2e3e3b0e66803122b5c4c7dd36d43ddStephen Hines mProxy->interrupt(); 541ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray } 542ce8b0e674c93035013d1c33aaabc9bb6ceffde0fTim Murray mProxy->flush(); 54398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mAudioTrack->flush(); 54498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams} 54598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 54695730ed847182c0b424f87d14d4ce276496dfa66Stephen Hinesvoid AudioTrack::pause() 54795730ed847182c0b424f87d14d4ce276496dfa66Stephen Hines{ 54895730ed847182c0b424f87d14d4ce276496dfa66Stephen Hines AutoMutex lock(mLock); 54995730ed847182c0b424f87d14d4ce276496dfa66Stephen Hines if (mState == STATE_ACTIVE) { 55095730ed847182c0b424f87d14d4ce276496dfa66Stephen Hines mState = STATE_PAUSED; 55195730ed847182c0b424f87d14d4ce276496dfa66Stephen Hines } else if (mState == STATE_STOPPING) { 55295730ed847182c0b424f87d14d4ce276496dfa66Stephen Hines mState = STATE_PAUSED_STOPPING; 55398a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } else { 55498a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams return; 55598a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams } 55698a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mProxy->interrupt(); 55798a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams mAudioTrack->pause(); 55898a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams 55998a281354fe06d1f970d0521c9a08d9eb0aa1a45Jason Sams if (isOffloaded_l()) { 560 if (mOutput != AUDIO_IO_HANDLE_NONE) { 561 uint32_t halFrames; 562 // OffloadThread sends HAL pause in its threadLoop.. time saved 563 // here can be slightly off 564 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 565 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 566 } 567 } 568} 569 570status_t AudioTrack::setVolume(float left, float right) 571{ 572 // This duplicates a test by AudioTrack JNI, but that is not the only caller 573 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 574 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 575 return BAD_VALUE; 576 } 577 578 AutoMutex lock(mLock); 579 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 580 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 581 582 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 583 584 if (isOffloaded_l()) { 585 mAudioTrack->signal(); 586 } 587 return NO_ERROR; 588} 589 590status_t AudioTrack::setVolume(float volume) 591{ 592 return setVolume(volume, volume); 593} 594 595status_t AudioTrack::setAuxEffectSendLevel(float level) 596{ 597 // This duplicates a test by AudioTrack JNI, but that is not the only caller 598 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 599 return BAD_VALUE; 600 } 601 602 AutoMutex lock(mLock); 603 mSendLevel = level; 604 mProxy->setSendLevel(level); 605 606 return NO_ERROR; 607} 608 609void AudioTrack::getAuxEffectSendLevel(float* level) const 610{ 611 if (level != NULL) { 612 *level = mSendLevel; 613 } 614} 615 616status_t AudioTrack::setSampleRate(uint32_t rate) 617{ 618 if (mIsTimed || isOffloaded()) { 619 return INVALID_OPERATION; 620 } 621 622 uint32_t afSamplingRate; 623 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 624 return NO_INIT; 625 } 626 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 627 if (rate == 0 || rate > afSamplingRate*2 ) { 628 return BAD_VALUE; 629 } 630 631 AutoMutex lock(mLock); 632 mSampleRate = rate; 633 mProxy->setSampleRate(rate); 634 635 return NO_ERROR; 636} 637 638uint32_t AudioTrack::getSampleRate() const 639{ 640 if (mIsTimed) { 641 return 0; 642 } 643 644 AutoMutex lock(mLock); 645 646 // sample rate can be updated during playback by the offloaded decoder so we need to 647 // query the HAL and update if needed. 648// FIXME use Proxy return channel to update the rate from server and avoid polling here 649 if (isOffloaded_l()) { 650 if (mOutput != AUDIO_IO_HANDLE_NONE) { 651 uint32_t sampleRate = 0; 652 status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); 653 if (status == NO_ERROR) { 654 mSampleRate = sampleRate; 655 } 656 } 657 } 658 return mSampleRate; 659} 660 661status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 662{ 663 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 664 return INVALID_OPERATION; 665 } 666 667 if (loopCount == 0) { 668 ; 669 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 670 loopEnd - loopStart >= MIN_LOOP) { 671 ; 672 } else { 673 return BAD_VALUE; 674 } 675 676 AutoMutex lock(mLock); 677 // See setPosition() regarding setting parameters such as loop points or position while active 678 if (mState == STATE_ACTIVE) { 679 return INVALID_OPERATION; 680 } 681 setLoop_l(loopStart, loopEnd, loopCount); 682 return NO_ERROR; 683} 684 685void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 686{ 687 // FIXME If setting a loop also sets position to start of loop, then 688 // this is correct. Otherwise it should be removed. 689 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 690 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 691 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 692} 693 694status_t AudioTrack::setMarkerPosition(uint32_t marker) 695{ 696 // The only purpose of setting marker position is to get a callback 697 if (mCbf == NULL || isOffloaded()) { 698 return INVALID_OPERATION; 699 } 700 701 AutoMutex lock(mLock); 702 mMarkerPosition = marker; 703 mMarkerReached = false; 704 705 return NO_ERROR; 706} 707 708status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 709{ 710 if (isOffloaded()) { 711 return INVALID_OPERATION; 712 } 713 if (marker == NULL) { 714 return BAD_VALUE; 715 } 716 717 AutoMutex lock(mLock); 718 *marker = mMarkerPosition; 719 720 return NO_ERROR; 721} 722 723status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 724{ 725 // The only purpose of setting position update period is to get a callback 726 if (mCbf == NULL || isOffloaded()) { 727 return INVALID_OPERATION; 728 } 729 730 AutoMutex lock(mLock); 731 mNewPosition = mProxy->getPosition() + updatePeriod; 732 mUpdatePeriod = updatePeriod; 733 734 return NO_ERROR; 735} 736 737status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 738{ 739 if (isOffloaded()) { 740 return INVALID_OPERATION; 741 } 742 if (updatePeriod == NULL) { 743 return BAD_VALUE; 744 } 745 746 AutoMutex lock(mLock); 747 *updatePeriod = mUpdatePeriod; 748 749 return NO_ERROR; 750} 751 752status_t AudioTrack::setPosition(uint32_t position) 753{ 754 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 755 return INVALID_OPERATION; 756 } 757 if (position > mFrameCount) { 758 return BAD_VALUE; 759 } 760 761 AutoMutex lock(mLock); 762 // Currently we require that the player is inactive before setting parameters such as position 763 // or loop points. Otherwise, there could be a race condition: the application could read the 764 // current position, compute a new position or loop parameters, and then set that position or 765 // loop parameters but it would do the "wrong" thing since the position has continued to advance 766 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 767 // to specify how it wants to handle such scenarios. 768 if (mState == STATE_ACTIVE) { 769 return INVALID_OPERATION; 770 } 771 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 772 mLoopPeriod = 0; 773 // FIXME Check whether loops and setting position are incompatible in old code. 774 // If we use setLoop for both purposes we lose the capability to set the position while looping. 775 mStaticProxy->setLoop(position, mFrameCount, 0); 776 777 return NO_ERROR; 778} 779 780status_t AudioTrack::getPosition(uint32_t *position) const 781{ 782 if (position == NULL) { 783 return BAD_VALUE; 784 } 785 786 AutoMutex lock(mLock); 787 if (isOffloaded_l()) { 788 uint32_t dspFrames = 0; 789 790 if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) { 791 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 792 *position = mPausedPosition; 793 return NO_ERROR; 794 } 795 796 if (mOutput != AUDIO_IO_HANDLE_NONE) { 797 uint32_t halFrames; 798 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 799 } 800 *position = dspFrames; 801 } else { 802 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 803 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 804 mProxy->getPosition(); 805 } 806 return NO_ERROR; 807} 808 809status_t AudioTrack::getBufferPosition(uint32_t *position) 810{ 811 if (mSharedBuffer == 0 || mIsTimed) { 812 return INVALID_OPERATION; 813 } 814 if (position == NULL) { 815 return BAD_VALUE; 816 } 817 818 AutoMutex lock(mLock); 819 *position = mStaticProxy->getBufferPosition(); 820 return NO_ERROR; 821} 822 823status_t AudioTrack::reload() 824{ 825 if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { 826 return INVALID_OPERATION; 827 } 828 829 AutoMutex lock(mLock); 830 // See setPosition() regarding setting parameters such as loop points or position while active 831 if (mState == STATE_ACTIVE) { 832 return INVALID_OPERATION; 833 } 834 mNewPosition = mUpdatePeriod; 835 mLoopPeriod = 0; 836 // FIXME The new code cannot reload while keeping a loop specified. 837 // Need to check how the old code handled this, and whether it's a significant change. 838 mStaticProxy->setLoop(0, mFrameCount, 0); 839 return NO_ERROR; 840} 841 842audio_io_handle_t AudioTrack::getOutput() const 843{ 844 AutoMutex lock(mLock); 845 return mOutput; 846} 847 848status_t AudioTrack::attachAuxEffect(int effectId) 849{ 850 AutoMutex lock(mLock); 851 status_t status = mAudioTrack->attachAuxEffect(effectId); 852 if (status == NO_ERROR) { 853 mAuxEffectId = effectId; 854 } 855 return status; 856} 857 858// ------------------------------------------------------------------------- 859 860// must be called with mLock held 861status_t AudioTrack::createTrack_l(size_t epoch) 862{ 863 status_t status; 864 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 865 if (audioFlinger == 0) { 866 ALOGE("Could not get audioflinger"); 867 return NO_INIT; 868 } 869 870 audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat, 871 mChannelMask, mFlags, mOffloadInfo); 872 if (output == AUDIO_IO_HANDLE_NONE) { 873 ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, " 874 "channel mask %#x, flags %#x", 875 mStreamType, mSampleRate, mFormat, mChannelMask, mFlags); 876 return BAD_VALUE; 877 } 878 { 879 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 880 // we must release it ourselves if anything goes wrong. 881 882 // Not all of these values are needed under all conditions, but it is easier to get them all 883 884 uint32_t afLatency; 885 status = AudioSystem::getLatency(output, &afLatency); 886 if (status != NO_ERROR) { 887 ALOGE("getLatency(%d) failed status %d", output, status); 888 goto release; 889 } 890 891 size_t afFrameCount; 892 status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount); 893 if (status != NO_ERROR) { 894 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status); 895 goto release; 896 } 897 898 uint32_t afSampleRate; 899 status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate); 900 if (status != NO_ERROR) { 901 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status); 902 goto release; 903 } 904 905 // Client decides whether the track is TIMED (see below), but can only express a preference 906 // for FAST. Server will perform additional tests. 907 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 908 // either of these use cases: 909 // use case 1: shared buffer 910 (mSharedBuffer != 0) || 911 // use case 2: callback transfer mode 912 (mTransfer == TRANSFER_CALLBACK)) && 913 // matching sample rate 914 (mSampleRate == afSampleRate))) { 915 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 916 // once denied, do not request again if IAudioTrack is re-created 917 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 918 } 919 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 920 921 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 922 // n = 1 fast track with single buffering; nBuffering is ignored 923 // n = 2 fast track with double buffering 924 // n = 2 normal track, no sample rate conversion 925 // n = 3 normal track, with sample rate conversion 926 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 927 // n > 3 very high latency or very small notification interval; nBuffering is ignored 928 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 929 930 mNotificationFramesAct = mNotificationFramesReq; 931 932 size_t frameCount = mReqFrameCount; 933 if (!audio_is_linear_pcm(mFormat)) { 934 935 if (mSharedBuffer != 0) { 936 // Same comment as below about ignoring frameCount parameter for set() 937 frameCount = mSharedBuffer->size(); 938 } else if (frameCount == 0) { 939 frameCount = afFrameCount; 940 } 941 if (mNotificationFramesAct != frameCount) { 942 mNotificationFramesAct = frameCount; 943 } 944 } else if (mSharedBuffer != 0) { 945 946 // Ensure that buffer alignment matches channel count 947 // 8-bit data in shared memory is not currently supported by AudioFlinger 948 size_t alignment = audio_bytes_per_sample( 949 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 950 if (alignment & 1) { 951 alignment = 1; 952 } 953 if (mChannelCount > 1) { 954 // More than 2 channels does not require stronger alignment than stereo 955 alignment <<= 1; 956 } 957 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 958 ALOGE("Invalid buffer alignment: address %p, channel count %u", 959 mSharedBuffer->pointer(), mChannelCount); 960 status = BAD_VALUE; 961 goto release; 962 } 963 964 // When initializing a shared buffer AudioTrack via constructors, 965 // there's no frameCount parameter. 966 // But when initializing a shared buffer AudioTrack via set(), 967 // there _is_ a frameCount parameter. We silently ignore it. 968 frameCount = mSharedBuffer->size() / mFrameSizeAF; 969 970 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 971 972 // FIXME move these calculations and associated checks to server 973 974 // Ensure that buffer depth covers at least audio hardware latency 975 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 976 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 977 afFrameCount, minBufCount, afSampleRate, afLatency); 978 if (minBufCount <= nBuffering) { 979 minBufCount = nBuffering; 980 } 981 982 size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate; 983 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 984 ", afLatency=%d", 985 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 986 987 if (frameCount == 0) { 988 frameCount = minFrameCount; 989 } else if (frameCount < minFrameCount) { 990 // not ALOGW because it happens all the time when playing key clicks over A2DP 991 ALOGV("Minimum buffer size corrected from %d to %d", 992 frameCount, minFrameCount); 993 frameCount = minFrameCount; 994 } 995 // Make sure that application is notified with sufficient margin before underrun 996 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 997 mNotificationFramesAct = frameCount/nBuffering; 998 } 999 1000 } else { 1001 // For fast tracks, the frame count calculations and checks are done by server 1002 } 1003 1004 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1005 if (mIsTimed) { 1006 trackFlags |= IAudioFlinger::TRACK_TIMED; 1007 } 1008 1009 pid_t tid = -1; 1010 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1011 trackFlags |= IAudioFlinger::TRACK_FAST; 1012 if (mAudioTrackThread != 0) { 1013 tid = mAudioTrackThread->getTid(); 1014 } 1015 } 1016 1017 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1018 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1019 } 1020 1021 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1022 // but we will still need the original value also 1023 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1024 mSampleRate, 1025 // AudioFlinger only sees 16-bit PCM 1026 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1027 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1028 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1029 mChannelMask, 1030 &temp, 1031 &trackFlags, 1032 mSharedBuffer, 1033 output, 1034 tid, 1035 &mSessionId, 1036 mClientUid, 1037 &status); 1038 1039 if (status != NO_ERROR) { 1040 ALOGE("AudioFlinger could not create track, status: %d", status); 1041 goto release; 1042 } 1043 ALOG_ASSERT(track != 0); 1044 1045 // AudioFlinger now owns the reference to the I/O handle, 1046 // so we are no longer responsible for releasing it. 1047 1048 sp<IMemory> iMem = track->getCblk(); 1049 if (iMem == 0) { 1050 ALOGE("Could not get control block"); 1051 return NO_INIT; 1052 } 1053 void *iMemPointer = iMem->pointer(); 1054 if (iMemPointer == NULL) { 1055 ALOGE("Could not get control block pointer"); 1056 return NO_INIT; 1057 } 1058 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1059 if (mAudioTrack != 0) { 1060 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1061 mDeathNotifier.clear(); 1062 } 1063 mAudioTrack = track; 1064 mCblkMemory = iMem; 1065 IPCThreadState::self()->flushCommands(); 1066 1067 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1068 mCblk = cblk; 1069 // note that temp is the (possibly revised) value of frameCount 1070 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1071 // In current design, AudioTrack client checks and ensures frame count validity before 1072 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1073 // for fast track as it uses a special method of assigning frame count. 1074 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 1075 } 1076 frameCount = temp; 1077 1078 mAwaitBoost = false; 1079 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1080 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1081 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 1082 mAwaitBoost = true; 1083 if (mSharedBuffer == 0) { 1084 // Theoretically double-buffering is not required for fast tracks, 1085 // due to tighter scheduling. But in practice, to accommodate kernels with 1086 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1087 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1088 mNotificationFramesAct = frameCount/nBuffering; 1089 } 1090 } 1091 } else { 1092 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 1093 // once denied, do not request again if IAudioTrack is re-created 1094 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1095 if (mSharedBuffer == 0) { 1096 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1097 mNotificationFramesAct = frameCount/nBuffering; 1098 } 1099 } 1100 } 1101 } 1102 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1103 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1104 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1105 } else { 1106 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1107 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1108 // FIXME This is a warning, not an error, so don't return error status 1109 //return NO_INIT; 1110 } 1111 } 1112 1113 // We retain a copy of the I/O handle, but don't own the reference 1114 mOutput = output; 1115 mRefreshRemaining = true; 1116 1117 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1118 // is the value of pointer() for the shared buffer, otherwise buffers points 1119 // immediately after the control block. This address is for the mapping within client 1120 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1121 void* buffers; 1122 if (mSharedBuffer == 0) { 1123 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1124 } else { 1125 buffers = mSharedBuffer->pointer(); 1126 } 1127 1128 mAudioTrack->attachAuxEffect(mAuxEffectId); 1129 // FIXME don't believe this lie 1130 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1131 1132 mFrameCount = frameCount; 1133 // If IAudioTrack is re-created, don't let the requested frameCount 1134 // decrease. This can confuse clients that cache frameCount(). 1135 if (frameCount > mReqFrameCount) { 1136 mReqFrameCount = frameCount; 1137 } 1138 1139 // update proxy 1140 if (mSharedBuffer == 0) { 1141 mStaticProxy.clear(); 1142 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1143 } else { 1144 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1145 mProxy = mStaticProxy; 1146 } 1147 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1148 mProxy->setSendLevel(mSendLevel); 1149 mProxy->setSampleRate(mSampleRate); 1150 mProxy->setEpoch(epoch); 1151 mProxy->setMinimum(mNotificationFramesAct); 1152 1153 mDeathNotifier = new DeathNotifier(this); 1154 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1155 1156 return NO_ERROR; 1157 } 1158 1159release: 1160 AudioSystem::releaseOutput(output); 1161 if (status == NO_ERROR) { 1162 status = NO_INIT; 1163 } 1164 return status; 1165} 1166 1167status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1168{ 1169 if (audioBuffer == NULL) { 1170 return BAD_VALUE; 1171 } 1172 if (mTransfer != TRANSFER_OBTAIN) { 1173 audioBuffer->frameCount = 0; 1174 audioBuffer->size = 0; 1175 audioBuffer->raw = NULL; 1176 return INVALID_OPERATION; 1177 } 1178 1179 const struct timespec *requested; 1180 struct timespec timeout; 1181 if (waitCount == -1) { 1182 requested = &ClientProxy::kForever; 1183 } else if (waitCount == 0) { 1184 requested = &ClientProxy::kNonBlocking; 1185 } else if (waitCount > 0) { 1186 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1187 timeout.tv_sec = ms / 1000; 1188 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1189 requested = &timeout; 1190 } else { 1191 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1192 requested = NULL; 1193 } 1194 return obtainBuffer(audioBuffer, requested); 1195} 1196 1197status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1198 struct timespec *elapsed, size_t *nonContig) 1199{ 1200 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1201 uint32_t oldSequence = 0; 1202 uint32_t newSequence; 1203 1204 Proxy::Buffer buffer; 1205 status_t status = NO_ERROR; 1206 1207 static const int32_t kMaxTries = 5; 1208 int32_t tryCounter = kMaxTries; 1209 1210 do { 1211 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1212 // keep them from going away if another thread re-creates the track during obtainBuffer() 1213 sp<AudioTrackClientProxy> proxy; 1214 sp<IMemory> iMem; 1215 1216 { // start of lock scope 1217 AutoMutex lock(mLock); 1218 1219 newSequence = mSequence; 1220 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1221 if (status == DEAD_OBJECT) { 1222 // re-create track, unless someone else has already done so 1223 if (newSequence == oldSequence) { 1224 status = restoreTrack_l("obtainBuffer"); 1225 if (status != NO_ERROR) { 1226 buffer.mFrameCount = 0; 1227 buffer.mRaw = NULL; 1228 buffer.mNonContig = 0; 1229 break; 1230 } 1231 } 1232 } 1233 oldSequence = newSequence; 1234 1235 // Keep the extra references 1236 proxy = mProxy; 1237 iMem = mCblkMemory; 1238 1239 if (mState == STATE_STOPPING) { 1240 status = -EINTR; 1241 buffer.mFrameCount = 0; 1242 buffer.mRaw = NULL; 1243 buffer.mNonContig = 0; 1244 break; 1245 } 1246 1247 // Non-blocking if track is stopped or paused 1248 if (mState != STATE_ACTIVE) { 1249 requested = &ClientProxy::kNonBlocking; 1250 } 1251 1252 } // end of lock scope 1253 1254 buffer.mFrameCount = audioBuffer->frameCount; 1255 // FIXME starts the requested timeout and elapsed over from scratch 1256 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1257 1258 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1259 1260 audioBuffer->frameCount = buffer.mFrameCount; 1261 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1262 audioBuffer->raw = buffer.mRaw; 1263 if (nonContig != NULL) { 1264 *nonContig = buffer.mNonContig; 1265 } 1266 return status; 1267} 1268 1269void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1270{ 1271 if (mTransfer == TRANSFER_SHARED) { 1272 return; 1273 } 1274 1275 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1276 if (stepCount == 0) { 1277 return; 1278 } 1279 1280 Proxy::Buffer buffer; 1281 buffer.mFrameCount = stepCount; 1282 buffer.mRaw = audioBuffer->raw; 1283 1284 AutoMutex lock(mLock); 1285 mInUnderrun = false; 1286 mProxy->releaseBuffer(&buffer); 1287 1288 // restart track if it was disabled by audioflinger due to previous underrun 1289 if (mState == STATE_ACTIVE) { 1290 audio_track_cblk_t* cblk = mCblk; 1291 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1292 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1293 // FIXME ignoring status 1294 mAudioTrack->start(); 1295 } 1296 } 1297} 1298 1299// ------------------------------------------------------------------------- 1300 1301ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1302{ 1303 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1304 return INVALID_OPERATION; 1305 } 1306 1307 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1308 // Sanity-check: user is most-likely passing an error code, and it would 1309 // make the return value ambiguous (actualSize vs error). 1310 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1311 return BAD_VALUE; 1312 } 1313 1314 size_t written = 0; 1315 Buffer audioBuffer; 1316 1317 while (userSize >= mFrameSize) { 1318 audioBuffer.frameCount = userSize / mFrameSize; 1319 1320 status_t err = obtainBuffer(&audioBuffer, 1321 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1322 if (err < 0) { 1323 if (written > 0) { 1324 break; 1325 } 1326 return ssize_t(err); 1327 } 1328 1329 size_t toWrite; 1330 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1331 // Divide capacity by 2 to take expansion into account 1332 toWrite = audioBuffer.size >> 1; 1333 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1334 } else { 1335 toWrite = audioBuffer.size; 1336 memcpy(audioBuffer.i8, buffer, toWrite); 1337 } 1338 buffer = ((const char *) buffer) + toWrite; 1339 userSize -= toWrite; 1340 written += toWrite; 1341 1342 releaseBuffer(&audioBuffer); 1343 } 1344 1345 return written; 1346} 1347 1348// ------------------------------------------------------------------------- 1349 1350TimedAudioTrack::TimedAudioTrack() { 1351 mIsTimed = true; 1352} 1353 1354status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1355{ 1356 AutoMutex lock(mLock); 1357 status_t result = UNKNOWN_ERROR; 1358 1359#if 1 1360 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1361 // while we are accessing the cblk 1362 sp<IAudioTrack> audioTrack = mAudioTrack; 1363 sp<IMemory> iMem = mCblkMemory; 1364#endif 1365 1366 // If the track is not invalid already, try to allocate a buffer. alloc 1367 // fails indicating that the server is dead, flag the track as invalid so 1368 // we can attempt to restore in just a bit. 1369 audio_track_cblk_t* cblk = mCblk; 1370 if (!(cblk->mFlags & CBLK_INVALID)) { 1371 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1372 if (result == DEAD_OBJECT) { 1373 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1374 } 1375 } 1376 1377 // If the track is invalid at this point, attempt to restore it. and try the 1378 // allocation one more time. 1379 if (cblk->mFlags & CBLK_INVALID) { 1380 result = restoreTrack_l("allocateTimedBuffer"); 1381 1382 if (result == NO_ERROR) { 1383 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1384 } 1385 } 1386 1387 return result; 1388} 1389 1390status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1391 int64_t pts) 1392{ 1393 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1394 { 1395 AutoMutex lock(mLock); 1396 audio_track_cblk_t* cblk = mCblk; 1397 // restart track if it was disabled by audioflinger due to previous underrun 1398 if (buffer->size() != 0 && status == NO_ERROR && 1399 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1400 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1401 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1402 // FIXME ignoring status 1403 mAudioTrack->start(); 1404 } 1405 } 1406 return status; 1407} 1408 1409status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1410 TargetTimeline target) 1411{ 1412 return mAudioTrack->setMediaTimeTransform(xform, target); 1413} 1414 1415// ------------------------------------------------------------------------- 1416 1417nsecs_t AudioTrack::processAudioBuffer() 1418{ 1419 // Currently the AudioTrack thread is not created if there are no callbacks. 1420 // Would it ever make sense to run the thread, even without callbacks? 1421 // If so, then replace this by checks at each use for mCbf != NULL. 1422 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1423 1424 mLock.lock(); 1425 if (mAwaitBoost) { 1426 mAwaitBoost = false; 1427 mLock.unlock(); 1428 static const int32_t kMaxTries = 5; 1429 int32_t tryCounter = kMaxTries; 1430 uint32_t pollUs = 10000; 1431 do { 1432 int policy = sched_getscheduler(0); 1433 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1434 break; 1435 } 1436 usleep(pollUs); 1437 pollUs <<= 1; 1438 } while (tryCounter-- > 0); 1439 if (tryCounter < 0) { 1440 ALOGE("did not receive expected priority boost on time"); 1441 } 1442 // Run again immediately 1443 return 0; 1444 } 1445 1446 // Can only reference mCblk while locked 1447 int32_t flags = android_atomic_and( 1448 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1449 1450 // Check for track invalidation 1451 if (flags & CBLK_INVALID) { 1452 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1453 // AudioSystem cache. We should not exit here but after calling the callback so 1454 // that the upper layers can recreate the track 1455 if (!isOffloaded_l() || (mSequence == mObservedSequence)) { 1456 status_t status = restoreTrack_l("processAudioBuffer"); 1457 mLock.unlock(); 1458 // Run again immediately, but with a new IAudioTrack 1459 return 0; 1460 } 1461 } 1462 1463 bool waitStreamEnd = mState == STATE_STOPPING; 1464 bool active = mState == STATE_ACTIVE; 1465 1466 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1467 bool newUnderrun = false; 1468 if (flags & CBLK_UNDERRUN) { 1469#if 0 1470 // Currently in shared buffer mode, when the server reaches the end of buffer, 1471 // the track stays active in continuous underrun state. It's up to the application 1472 // to pause or stop the track, or set the position to a new offset within buffer. 1473 // This was some experimental code to auto-pause on underrun. Keeping it here 1474 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1475 if (mTransfer == TRANSFER_SHARED) { 1476 mState = STATE_PAUSED; 1477 active = false; 1478 } 1479#endif 1480 if (!mInUnderrun) { 1481 mInUnderrun = true; 1482 newUnderrun = true; 1483 } 1484 } 1485 1486 // Get current position of server 1487 size_t position = mProxy->getPosition(); 1488 1489 // Manage marker callback 1490 bool markerReached = false; 1491 size_t markerPosition = mMarkerPosition; 1492 // FIXME fails for wraparound, need 64 bits 1493 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1494 mMarkerReached = markerReached = true; 1495 } 1496 1497 // Determine number of new position callback(s) that will be needed, while locked 1498 size_t newPosCount = 0; 1499 size_t newPosition = mNewPosition; 1500 size_t updatePeriod = mUpdatePeriod; 1501 // FIXME fails for wraparound, need 64 bits 1502 if (updatePeriod > 0 && position >= newPosition) { 1503 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1504 mNewPosition += updatePeriod * newPosCount; 1505 } 1506 1507 // Cache other fields that will be needed soon 1508 uint32_t loopPeriod = mLoopPeriod; 1509 uint32_t sampleRate = mSampleRate; 1510 uint32_t notificationFrames = mNotificationFramesAct; 1511 if (mRefreshRemaining) { 1512 mRefreshRemaining = false; 1513 mRemainingFrames = notificationFrames; 1514 mRetryOnPartialBuffer = false; 1515 } 1516 size_t misalignment = mProxy->getMisalignment(); 1517 uint32_t sequence = mSequence; 1518 sp<AudioTrackClientProxy> proxy = mProxy; 1519 1520 // These fields don't need to be cached, because they are assigned only by set(): 1521 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1522 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1523 1524 mLock.unlock(); 1525 1526 if (waitStreamEnd) { 1527 struct timespec timeout; 1528 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1529 timeout.tv_nsec = 0; 1530 1531 status_t status = proxy->waitStreamEndDone(&timeout); 1532 switch (status) { 1533 case NO_ERROR: 1534 case DEAD_OBJECT: 1535 case TIMED_OUT: 1536 mCbf(EVENT_STREAM_END, mUserData, NULL); 1537 { 1538 AutoMutex lock(mLock); 1539 // The previously assigned value of waitStreamEnd is no longer valid, 1540 // since the mutex has been unlocked and either the callback handler 1541 // or another thread could have re-started the AudioTrack during that time. 1542 waitStreamEnd = mState == STATE_STOPPING; 1543 if (waitStreamEnd) { 1544 mState = STATE_STOPPED; 1545 } 1546 } 1547 if (waitStreamEnd && status != DEAD_OBJECT) { 1548 return NS_INACTIVE; 1549 } 1550 break; 1551 } 1552 return 0; 1553 } 1554 1555 // perform callbacks while unlocked 1556 if (newUnderrun) { 1557 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1558 } 1559 // FIXME we will miss loops if loop cycle was signaled several times since last call 1560 // to processAudioBuffer() 1561 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1562 mCbf(EVENT_LOOP_END, mUserData, NULL); 1563 } 1564 if (flags & CBLK_BUFFER_END) { 1565 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1566 } 1567 if (markerReached) { 1568 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1569 } 1570 while (newPosCount > 0) { 1571 size_t temp = newPosition; 1572 mCbf(EVENT_NEW_POS, mUserData, &temp); 1573 newPosition += updatePeriod; 1574 newPosCount--; 1575 } 1576 1577 if (mObservedSequence != sequence) { 1578 mObservedSequence = sequence; 1579 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1580 // for offloaded tracks, just wait for the upper layers to recreate the track 1581 if (isOffloaded()) { 1582 return NS_INACTIVE; 1583 } 1584 } 1585 1586 // if inactive, then don't run me again until re-started 1587 if (!active) { 1588 return NS_INACTIVE; 1589 } 1590 1591 // Compute the estimated time until the next timed event (position, markers, loops) 1592 // FIXME only for non-compressed audio 1593 uint32_t minFrames = ~0; 1594 if (!markerReached && position < markerPosition) { 1595 minFrames = markerPosition - position; 1596 } 1597 if (loopPeriod > 0 && loopPeriod < minFrames) { 1598 minFrames = loopPeriod; 1599 } 1600 if (updatePeriod > 0 && updatePeriod < minFrames) { 1601 minFrames = updatePeriod; 1602 } 1603 1604 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1605 static const uint32_t kPoll = 0; 1606 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1607 minFrames = kPoll * notificationFrames; 1608 } 1609 1610 // Convert frame units to time units 1611 nsecs_t ns = NS_WHENEVER; 1612 if (minFrames != (uint32_t) ~0) { 1613 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1614 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1615 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1616 } 1617 1618 // If not supplying data by EVENT_MORE_DATA, then we're done 1619 if (mTransfer != TRANSFER_CALLBACK) { 1620 return ns; 1621 } 1622 1623 struct timespec timeout; 1624 const struct timespec *requested = &ClientProxy::kForever; 1625 if (ns != NS_WHENEVER) { 1626 timeout.tv_sec = ns / 1000000000LL; 1627 timeout.tv_nsec = ns % 1000000000LL; 1628 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1629 requested = &timeout; 1630 } 1631 1632 while (mRemainingFrames > 0) { 1633 1634 Buffer audioBuffer; 1635 audioBuffer.frameCount = mRemainingFrames; 1636 size_t nonContig; 1637 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1638 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1639 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1640 requested = &ClientProxy::kNonBlocking; 1641 size_t avail = audioBuffer.frameCount + nonContig; 1642 ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", 1643 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1644 if (err != NO_ERROR) { 1645 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1646 (isOffloaded() && (err == DEAD_OBJECT))) { 1647 return 0; 1648 } 1649 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1650 return NS_NEVER; 1651 } 1652 1653 if (mRetryOnPartialBuffer && !isOffloaded()) { 1654 mRetryOnPartialBuffer = false; 1655 if (avail < mRemainingFrames) { 1656 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1657 if (ns < 0 || myns < ns) { 1658 ns = myns; 1659 } 1660 return ns; 1661 } 1662 } 1663 1664 // Divide buffer size by 2 to take into account the expansion 1665 // due to 8 to 16 bit conversion: the callback must fill only half 1666 // of the destination buffer 1667 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1668 audioBuffer.size >>= 1; 1669 } 1670 1671 size_t reqSize = audioBuffer.size; 1672 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1673 size_t writtenSize = audioBuffer.size; 1674 1675 // Sanity check on returned size 1676 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1677 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1678 reqSize, (int) writtenSize); 1679 return NS_NEVER; 1680 } 1681 1682 if (writtenSize == 0) { 1683 // The callback is done filling buffers 1684 // Keep this thread going to handle timed events and 1685 // still try to get more data in intervals of WAIT_PERIOD_MS 1686 // but don't just loop and block the CPU, so wait 1687 return WAIT_PERIOD_MS * 1000000LL; 1688 } 1689 1690 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1691 // 8 to 16 bit conversion, note that source and destination are the same address 1692 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1693 audioBuffer.size <<= 1; 1694 } 1695 1696 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1697 audioBuffer.frameCount = releasedFrames; 1698 mRemainingFrames -= releasedFrames; 1699 if (misalignment >= releasedFrames) { 1700 misalignment -= releasedFrames; 1701 } else { 1702 misalignment = 0; 1703 } 1704 1705 releaseBuffer(&audioBuffer); 1706 1707 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1708 // if callback doesn't like to accept the full chunk 1709 if (writtenSize < reqSize) { 1710 continue; 1711 } 1712 1713 // There could be enough non-contiguous frames available to satisfy the remaining request 1714 if (mRemainingFrames <= nonContig) { 1715 continue; 1716 } 1717 1718#if 0 1719 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1720 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1721 // that total to a sum == notificationFrames. 1722 if (0 < misalignment && misalignment <= mRemainingFrames) { 1723 mRemainingFrames = misalignment; 1724 return (mRemainingFrames * 1100000000LL) / sampleRate; 1725 } 1726#endif 1727 1728 } 1729 mRemainingFrames = notificationFrames; 1730 mRetryOnPartialBuffer = true; 1731 1732 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1733 return 0; 1734} 1735 1736status_t AudioTrack::restoreTrack_l(const char *from) 1737{ 1738 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1739 isOffloaded_l() ? "Offloaded" : "PCM", from); 1740 ++mSequence; 1741 status_t result; 1742 1743 // refresh the audio configuration cache in this process to make sure we get new 1744 // output parameters in createTrack_l() 1745 AudioSystem::clearAudioConfigCache(); 1746 1747 if (isOffloaded_l()) { 1748 // FIXME re-creation of offloaded tracks is not yet implemented 1749 return DEAD_OBJECT; 1750 } 1751 1752 // if the new IAudioTrack is created, createTrack_l() will modify the 1753 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1754 // It will also delete the strong references on previous IAudioTrack and IMemory 1755 1756 // take the frames that will be lost by track recreation into account in saved position 1757 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1758 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1759 result = createTrack_l(position /*epoch*/); 1760 1761 if (result == NO_ERROR) { 1762 // continue playback from last known position, but 1763 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1764 if (mStaticProxy != NULL) { 1765 mLoopPeriod = 0; 1766 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1767 } 1768 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1769 // track destruction have been played? This is critical for SoundPool implementation 1770 // This must be broken, and needs to be tested/debugged. 1771#if 0 1772 // restore write index and set other indexes to reflect empty buffer status 1773 if (!strcmp(from, "start")) { 1774 // Make sure that a client relying on callback events indicating underrun or 1775 // the actual amount of audio frames played (e.g SoundPool) receives them. 1776 if (mSharedBuffer == 0) { 1777 // restart playback even if buffer is not completely filled. 1778 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1779 } 1780 } 1781#endif 1782 if (mState == STATE_ACTIVE) { 1783 result = mAudioTrack->start(); 1784 } 1785 } 1786 if (result != NO_ERROR) { 1787 ALOGW("restoreTrack_l() failed status %d", result); 1788 mState = STATE_STOPPED; 1789 } 1790 1791 return result; 1792} 1793 1794status_t AudioTrack::setParameters(const String8& keyValuePairs) 1795{ 1796 AutoMutex lock(mLock); 1797 return mAudioTrack->setParameters(keyValuePairs); 1798} 1799 1800status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1801{ 1802 AutoMutex lock(mLock); 1803 // FIXME not implemented for fast tracks; should use proxy and SSQ 1804 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1805 return INVALID_OPERATION; 1806 } 1807 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1808 return INVALID_OPERATION; 1809 } 1810 status_t status = mAudioTrack->getTimestamp(timestamp); 1811 if (status == NO_ERROR) { 1812 timestamp.mPosition += mProxy->getEpoch(); 1813 } 1814 return status; 1815} 1816 1817String8 AudioTrack::getParameters(const String8& keys) 1818{ 1819 audio_io_handle_t output = getOutput(); 1820 if (output != AUDIO_IO_HANDLE_NONE) { 1821 return AudioSystem::getParameters(output, keys); 1822 } else { 1823 return String8::empty(); 1824 } 1825} 1826 1827bool AudioTrack::isOffloaded() const 1828{ 1829 AutoMutex lock(mLock); 1830 return isOffloaded_l(); 1831} 1832 1833status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1834{ 1835 1836 const size_t SIZE = 256; 1837 char buffer[SIZE]; 1838 String8 result; 1839 1840 result.append(" AudioTrack::dump\n"); 1841 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1842 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 1843 result.append(buffer); 1844 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 1845 mChannelCount, mFrameCount); 1846 result.append(buffer); 1847 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1848 result.append(buffer); 1849 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1850 result.append(buffer); 1851 ::write(fd, result.string(), result.size()); 1852 return NO_ERROR; 1853} 1854 1855uint32_t AudioTrack::getUnderrunFrames() const 1856{ 1857 AutoMutex lock(mLock); 1858 return mProxy->getUnderrunFrames(); 1859} 1860 1861// ========================================================================= 1862 1863void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 1864{ 1865 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1866 if (audioTrack != 0) { 1867 AutoMutex lock(audioTrack->mLock); 1868 audioTrack->mProxy->binderDied(); 1869 } 1870} 1871 1872// ========================================================================= 1873 1874AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1875 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 1876 mIgnoreNextPausedInt(false) 1877{ 1878} 1879 1880AudioTrack::AudioTrackThread::~AudioTrackThread() 1881{ 1882} 1883 1884bool AudioTrack::AudioTrackThread::threadLoop() 1885{ 1886 { 1887 AutoMutex _l(mMyLock); 1888 if (mPaused) { 1889 mMyCond.wait(mMyLock); 1890 // caller will check for exitPending() 1891 return true; 1892 } 1893 if (mIgnoreNextPausedInt) { 1894 mIgnoreNextPausedInt = false; 1895 mPausedInt = false; 1896 } 1897 if (mPausedInt) { 1898 if (mPausedNs > 0) { 1899 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 1900 } else { 1901 mMyCond.wait(mMyLock); 1902 } 1903 mPausedInt = false; 1904 return true; 1905 } 1906 } 1907 nsecs_t ns = mReceiver.processAudioBuffer(); 1908 switch (ns) { 1909 case 0: 1910 return true; 1911 case NS_INACTIVE: 1912 pauseInternal(); 1913 return true; 1914 case NS_NEVER: 1915 return false; 1916 case NS_WHENEVER: 1917 // FIXME increase poll interval, or make event-driven 1918 ns = 1000000000LL; 1919 // fall through 1920 default: 1921 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1922 pauseInternal(ns); 1923 return true; 1924 } 1925} 1926 1927void AudioTrack::AudioTrackThread::requestExit() 1928{ 1929 // must be in this order to avoid a race condition 1930 Thread::requestExit(); 1931 resume(); 1932} 1933 1934void AudioTrack::AudioTrackThread::pause() 1935{ 1936 AutoMutex _l(mMyLock); 1937 mPaused = true; 1938} 1939 1940void AudioTrack::AudioTrackThread::resume() 1941{ 1942 AutoMutex _l(mMyLock); 1943 mIgnoreNextPausedInt = true; 1944 if (mPaused || mPausedInt) { 1945 mPaused = false; 1946 mPausedInt = false; 1947 mMyCond.signal(); 1948 } 1949} 1950 1951void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 1952{ 1953 AutoMutex _l(mMyLock); 1954 mPausedInt = true; 1955 mPausedNs = ns; 1956} 1957 1958}; // namespace android 1959