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-rw-r--r--services/audioflinger/Threads.cpp817
1 files changed, 583 insertions, 234 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 51025fe..1a20fae 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -23,7 +23,9 @@
#include "Configuration.h"
#include <math.h>
#include <fcntl.h>
+#include <linux/futex.h>
#include <sys/stat.h>
+#include <sys/syscall.h>
#include <cutils/properties.h>
#include <media/AudioParameter.h>
#include <media/AudioResamplerPublic.h>
@@ -84,7 +86,13 @@
#define ALOGVV(a...) do { } while(0)
#endif
+// TODO: Move these macro/inlines to a header file.
#define max(a, b) ((a) > (b) ? (a) : (b))
+template <typename T>
+static inline T min(const T& a, const T& b)
+{
+ return a < b ? a : b;
+}
namespace android {
@@ -314,6 +322,165 @@ void CpuStats::sample(const String8 &title
// ThreadBase
// ----------------------------------------------------------------------------
+// static
+const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
+{
+ switch (type) {
+ case MIXER:
+ return "MIXER";
+ case DIRECT:
+ return "DIRECT";
+ case DUPLICATING:
+ return "DUPLICATING";
+ case RECORD:
+ return "RECORD";
+ case OFFLOAD:
+ return "OFFLOAD";
+ default:
+ return "unknown";
+ }
+}
+
+String8 devicesToString(audio_devices_t devices)
+{
+ static const struct mapping {
+ audio_devices_t mDevices;
+ const char * mString;
+ } mappingsOut[] = {
+ AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE",
+ AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER",
+ AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET",
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE",
+ AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX",
+ AUDIO_DEVICE_NONE, "NONE", // must be last
+ }, mappingsIn[] = {
+ AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC",
+ AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET",
+ AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL",
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX",
+ AUDIO_DEVICE_NONE, "NONE", // must be last
+ };
+ String8 result;
+ audio_devices_t allDevices = AUDIO_DEVICE_NONE;
+ const mapping *entry;
+ if (devices & AUDIO_DEVICE_BIT_IN) {
+ devices &= ~AUDIO_DEVICE_BIT_IN;
+ entry = mappingsIn;
+ } else {
+ entry = mappingsOut;
+ }
+ for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) {
+ allDevices = (audio_devices_t) (allDevices | entry->mDevices);
+ if (devices & entry->mDevices) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.append(entry->mString);
+ }
+ }
+ if (devices & ~allDevices) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.appendFormat("0x%X", devices & ~allDevices);
+ }
+ if (result.isEmpty()) {
+ result.append(entry->mString);
+ }
+ return result;
+}
+
+String8 inputFlagsToString(audio_input_flags_t flags)
+{
+ static const struct mapping {
+ audio_input_flags_t mFlag;
+ const char * mString;
+ } mappings[] = {
+ AUDIO_INPUT_FLAG_FAST, "FAST",
+ AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD",
+ AUDIO_INPUT_FLAG_NONE, "NONE", // must be last
+ };
+ String8 result;
+ audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE;
+ const mapping *entry;
+ for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) {
+ allFlags = (audio_input_flags_t) (allFlags | entry->mFlag);
+ if (flags & entry->mFlag) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.append(entry->mString);
+ }
+ }
+ if (flags & ~allFlags) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.appendFormat("0x%X", flags & ~allFlags);
+ }
+ if (result.isEmpty()) {
+ result.append(entry->mString);
+ }
+ return result;
+}
+
+String8 outputFlagsToString(audio_output_flags_t flags)
+{
+ static const struct mapping {
+ audio_output_flags_t mFlag;
+ const char * mString;
+ } mappings[] = {
+ AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT",
+ AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY",
+ AUDIO_OUTPUT_FLAG_FAST, "FAST",
+ AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER",
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD",
+ AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING",
+ AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC",
+ AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last
+ };
+ String8 result;
+ audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE;
+ const mapping *entry;
+ for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) {
+ allFlags = (audio_output_flags_t) (allFlags | entry->mFlag);
+ if (flags & entry->mFlag) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.append(entry->mString);
+ }
+ }
+ if (flags & ~allFlags) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.appendFormat("0x%X", flags & ~allFlags);
+ }
+ if (result.isEmpty()) {
+ result.append(entry->mString);
+ }
+ return result;
+}
+
+const char *sourceToString(audio_source_t source)
+{
+ switch (source) {
+ case AUDIO_SOURCE_DEFAULT: return "default";
+ case AUDIO_SOURCE_MIC: return "mic";
+ case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink";
+ case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink";
+ case AUDIO_SOURCE_VOICE_CALL: return "voice call";
+ case AUDIO_SOURCE_CAMCORDER: return "camcorder";
+ case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition";
+ case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication";
+ case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix";
+ case AUDIO_SOURCE_FM_TUNER: return "FM tuner";
+ case AUDIO_SOURCE_HOTWORD: return "hotword";
+ default: return "unknown";
+ }
+}
+
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
: Thread(false /*canCallJava*/),
@@ -577,20 +744,22 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u
bool locked = AudioFlinger::dumpTryLock(mLock);
if (!locked) {
- dprintf(fd, "thread %p maybe dead locked\n", this);
+ dprintf(fd, "thread %p may be deadlocked\n", this);
}
+ dprintf(fd, " Thread name: %s\n", mThreadName);
dprintf(fd, " I/O handle: %d\n", mId);
dprintf(fd, " TID: %d\n", getTid());
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
- dprintf(fd, " Sample rate: %u\n", mSampleRate);
+ dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
+ dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize);
- dprintf(fd, " Channel Count: %u\n", mChannelCount);
- dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask,
+ dprintf(fd, " Channel count: %u\n", mChannelCount);
+ dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask,
channelMaskToString(mChannelMask, mType != RECORD).string());
- dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
- dprintf(fd, " Frame size: %zu\n", mFrameSize);
+ dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
+ dprintf(fd, " Frame size: %zu bytes\n", mFrameSize);
dprintf(fd, " Pending config events:");
size_t numConfig = mConfigEvents.size();
if (numConfig) {
@@ -602,6 +771,9 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u
} else {
dprintf(fd, " none\n");
}
+ dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string());
+ dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string());
+ dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource));
if (locked) {
mLock.unlock();
@@ -635,19 +807,19 @@ void AudioFlinger::ThreadBase::acquireWakeLock(int uid)
String16 AudioFlinger::ThreadBase::getWakeLockTag()
{
switch (mType) {
- case MIXER:
- return String16("AudioMix");
- case DIRECT:
- return String16("AudioDirectOut");
- case DUPLICATING:
- return String16("AudioDup");
- case RECORD:
- return String16("AudioIn");
- case OFFLOAD:
- return String16("AudioOffload");
- default:
- ALOG_ASSERT(false);
- return String16("AudioUnknown");
+ case MIXER:
+ return String16("AudioMix");
+ case DIRECT:
+ return String16("AudioDirectOut");
+ case DUPLICATING:
+ return String16("AudioDup");
+ case RECORD:
+ return String16("AudioIn");
+ case OFFLOAD:
+ return String16("AudioOffload");
+ default:
+ ALOG_ASSERT(false);
+ return String16("AudioUnknown");
}
}
@@ -674,7 +846,7 @@ void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
if (status == NO_ERROR) {
mWakeLockToken = binder;
}
- ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+ ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status);
}
}
@@ -687,7 +859,7 @@ void AudioFlinger::ThreadBase::releaseWakeLock()
void AudioFlinger::ThreadBase::releaseWakeLock_l()
{
if (mWakeLockToken != 0) {
- ALOGV("releaseWakeLock_l() %s", mName);
+ ALOGV("releaseWakeLock_l() %s", mThreadName);
if (mPowerManager != 0) {
mPowerManager->releaseWakeLock(mWakeLockToken, 0,
true /* FIXME force oneway contrary to .aidl */);
@@ -708,7 +880,7 @@ void AudioFlinger::ThreadBase::getPowerManager_l() {
sp<IBinder> binder =
defaultServiceManager()->checkService(String16("power"));
if (binder == 0) {
- ALOGW("Thread %s cannot connect to the power manager service", mName);
+ ALOGW("Thread %s cannot connect to the power manager service", mThreadName);
} else {
mPowerManager = interface_cast<IPowerManager>(binder);
binder->linkToDeath(mDeathRecipient);
@@ -728,7 +900,7 @@ void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uid
status_t status;
status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(),
true /* FIXME force oneway contrary to .aidl */);
- ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+ ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status);
}
}
@@ -912,7 +1084,7 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
// mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo).
if (mType == DIRECT) {
ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s",
- desc->name, mName);
+ desc->name, mThreadName);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -936,7 +1108,8 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
case DUPLICATING:
case RECORD:
default:
- ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName);
+ ALOGW("createEffect_l() Cannot add global effect %s on thread %s",
+ desc->name, mThreadName);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -1201,8 +1374,8 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
// mLatchD, mLatchQ,
mLatchDValid(false), mLatchQValid(false)
{
- snprintf(mName, kNameLength, "AudioOut_%X", id);
- mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
+ snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
+ mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
// Assumes constructor is called by AudioFlinger with it's mLock held, but
// it would be safer to explicitly pass initial masterVolume/masterMute as
@@ -1315,7 +1488,10 @@ void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& ar
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
- dprintf(fd, "\nOutput thread %p:\n", this);
+ dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type()));
+
+ dumpBase(fd, args);
+
dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
dprintf(fd, " Total writes: %d\n", mNumWrites);
@@ -1326,15 +1502,17 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>&
dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
dprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
-
- dumpBase(fd, args);
+ AudioStreamOut *output = mOutput;
+ audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
+ String8 flagsAsString = outputFlagsToString(flags);
+ dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string());
}
// Thread virtuals
void AudioFlinger::PlaybackThread::onFirstRef()
{
- run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
+ run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
}
// ThreadBase virtuals
@@ -1378,9 +1556,10 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
(
(sharedBuffer != 0)
) ||
- // use case 2: callback handler and frame count is default or at least as large as HAL
+ // use case 2: frame count is default or at least as large as HAL
(
- (tid != -1) &&
+ // we formerly checked for a callback handler (non-0 tid),
+ // but that is no longer required for TRANSFER_OBTAIN mode
((frameCount == 0) ||
(frameCount >= mFrameCount))
)
@@ -1420,20 +1599,25 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
audio_is_linear_pcm(format),
channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
*flags &= ~IAudioFlinger::TRACK_FAST;
- // For compatibility with AudioTrack calculation, buffer depth is forced
- // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
- // This is probably too conservative, but legacy application code may depend on it.
- // If you change this calculation, also review the start threshold which is related.
+ }
+ }
+ // For normal PCM streaming tracks, update minimum frame count.
+ // For compatibility with AudioTrack calculation, buffer depth is forced
+ // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
+ // This is probably too conservative, but legacy application code may depend on it.
+ // If you change this calculation, also review the start threshold which is related.
+ if (!(*flags & IAudioFlinger::TRACK_FAST)
+ && audio_is_linear_pcm(format) && sharedBuffer == 0) {
uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
if (minBufCount < 2) {
minBufCount = 2;
}
- size_t minFrameCount = mNormalFrameCount * minBufCount;
- if (frameCount < minFrameCount) {
+ size_t minFrameCount =
+ minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate);
+ if (frameCount < minFrameCount) { // including frameCount == 0
frameCount = minFrameCount;
}
- }
}
*pFrameCount = frameCount;
@@ -1831,7 +2015,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
LOG_FATAL("HAL format %#x not supported for mixed output",
mFormat);
}
- mFrameSize = audio_stream_out_frame_size(mOutput->stream);
+ mFrameSize = mOutput->getFrameSize();
mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
mFrameCount = mBufferSize / mFrameSize;
if (mFrameCount & 15) {
@@ -1861,6 +2045,22 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
}
}
+ if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
+ // For best precision, we use float instead of the associated output
+ // device format (typically PCM 16 bit).
+
+ mFormat = AUDIO_FORMAT_PCM_FLOAT;
+ mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
+ mBufferSize = mFrameSize * mFrameCount;
+
+ // TODO: We currently use the associated output device channel mask and sample rate.
+ // (1) Perhaps use the ORed channel mask of all downstream MixerThreads
+ // (if a valid mask) to avoid premature downmix.
+ // (2) Perhaps use the maximum sample rate of all downstream MixerThreads
+ // instead of the output device sample rate to avoid loss of high frequency information.
+ // This may need to be updated as MixerThread/OutputTracks are added and not here.
+ }
+
// Calculate size of normal sink buffer relative to the HAL output buffer size
double multiplier = 1.0;
if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
@@ -1966,7 +2166,7 @@ status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, ui
} else {
status_t status;
uint32_t frames;
- status = mOutput->stream->get_render_position(mOutput->stream, &frames);
+ status = mOutput->getRenderPosition(&frames);
*dspFrames = (size_t)frames;
return status;
}
@@ -2008,13 +2208,13 @@ uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
}
-AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
+AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
{
Mutex::Autolock _l(mLock);
return mOutput;
}
-AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
+AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
{
Mutex::Autolock _l(mLock);
AudioStreamOut *output = mOutput;
@@ -2137,6 +2337,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
} else {
bytesWritten = framesWritten;
}
+ mLatchDValid = false;
status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp);
if (status == NO_ERROR) {
size_t totalFramesWritten = mNormalSink->framesWritten();
@@ -2159,8 +2360,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
}
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
- bytesWritten = mOutput->stream->write(mOutput->stream,
- (char *)mSinkBuffer + offset, mBytesRemaining);
+ bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
@@ -2640,7 +2840,9 @@ bool AudioFlinger::PlaybackThread::threadLoop()
}
} else {
+ ATRACE_BEGIN("sleep");
usleep(sleepTime);
+ ATRACE_END();
}
}
@@ -2711,8 +2913,7 @@ status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
if ((mType == OFFLOAD || mType == DIRECT)
&& mOutput != NULL && mOutput->stream->get_presentation_position) {
uint64_t position64;
- int ret = mOutput->stream->get_presentation_position(
- mOutput->stream, &position64, &timestamp.mTime);
+ int ret = mOutput->getPresentationPosition(&position64, &timestamp.mTime);
if (ret == 0) {
timestamp.mPosition = (uint32_t)position64;
return NO_ERROR;
@@ -2800,6 +3001,12 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud
mNormalFrameCount);
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+ if (type == DUPLICATING) {
+ // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
+ // (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
+ // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
+ return;
+ }
// create an NBAIO sink for the HAL output stream, and negotiate
mOutputSink = new AudioStreamOutSink(output->stream);
size_t numCounterOffers = 0;
@@ -2841,6 +3048,7 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud
NBAIO_Format format = mOutputSink->format();
NBAIO_Format origformat = format;
// adjust format to match that of the Fast Mixer
+ ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
format.mFormat = fastMixerFormat;
format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
@@ -3020,8 +3228,10 @@ ssize_t AudioFlinger::MixerThread::threadLoop_write()
#endif
}
state->mCommand = FastMixerState::MIX_WRITE;
+#ifdef FAST_THREAD_STATISTICS
mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
- FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
+ FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
+#endif
sq->end();
sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
if (kUseFastMixer == FastMixer_Dynamic) {
@@ -3083,7 +3293,7 @@ bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
void AudioFlinger::PlaybackThread::threadLoop_standby()
{
ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
- mOutput->stream->common.standby(&mOutput->stream->common);
+ mOutput->standby();
if (mUseAsyncWrite != 0) {
// discard any pending drain or write ack by incrementing sequence
mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
@@ -3386,8 +3596,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
if (sr == mSampleRate) {
desiredFrames = mNormalFrameCount;
} else {
- // +1 for rounding and +1 for additional sample needed for interpolation
- desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
+ desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate);
// add frames already consumed but not yet released by the resampler
// because mAudioTrackServerProxy->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
@@ -3405,6 +3614,23 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
}
size_t framesReady = track->framesReady();
+ if (ATRACE_ENABLED()) {
+ // I wish we had formatted trace names
+ char traceName[16];
+ strcpy(traceName, "nRdy");
+ int name = track->name();
+ if (AudioMixer::TRACK0 <= name &&
+ name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) {
+ name -= AudioMixer::TRACK0;
+ traceName[4] = (name / 10) + '0';
+ traceName[5] = (name % 10) + '0';
+ } else {
+ traceName[4] = '?';
+ traceName[5] = '?';
+ }
+ traceName[6] = '\0';
+ ATRACE_INT(traceName, framesReady);
+ }
if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
@@ -3836,7 +4062,7 @@ bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePa
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
+ mOutput->standby();
mStandby = true;
mBytesWritten = 0;
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
@@ -4096,6 +4322,10 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep
}
if (track->isStopping_1()) {
track->mState = TrackBase::STOPPING_2;
+ if (last && mHwPaused) {
+ doHwResume = true;
+ mHwPaused = false;
+ }
}
if ((track->sharedBuffer() != 0) || track->isStopped() ||
track->isStopping_2() || track->isPaused()) {
@@ -4178,8 +4408,8 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix()
while (frameCount) {
AudioBufferProvider::Buffer buffer;
buffer.frameCount = frameCount;
- mActiveTrack->getNextBuffer(&buffer);
- if (buffer.raw == NULL) {
+ status_t status = mActiveTrack->getNextBuffer(&buffer);
+ if (status != NO_ERROR || buffer.raw == NULL) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
@@ -4235,14 +4465,17 @@ void AudioFlinger::DirectOutputThread::threadLoop_exit()
bool AudioFlinger::DirectOutputThread::shouldStandby_l()
{
bool trackPaused = false;
+ bool trackStopped = false;
// do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
// after a timeout and we will enter standby then.
if (mTracks.size() > 0) {
trackPaused = mTracks[mTracks.size() - 1]->isPaused();
+ trackStopped = mTracks[mTracks.size() - 1]->isStopped() ||
+ mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE;
}
- return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused));
+ return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped));
}
// getTrackName_l() must be called with ThreadBase::mLock held
@@ -4291,7 +4524,7 @@ bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& key
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
+ mOutput->standby();
mStandby = true;
mBytesWritten = 0;
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
@@ -4345,7 +4578,10 @@ void AudioFlinger::DirectOutputThread::cacheParameters_l()
// use shorter standby delay as on normal output to release
// hardware resources as soon as possible
- if (audio_is_linear_pcm(mFormat)) {
+ // no delay on outputs with HW A/V sync
+ if (usesHwAvSync()) {
+ standbyDelay = 0;
+ } else if (audio_is_linear_pcm(mFormat)) {
standbyDelay = microseconds(activeSleepTime*2);
} else {
standbyDelay = kOffloadStandbyDelayNs;
@@ -4354,9 +4590,7 @@ void AudioFlinger::DirectOutputThread::cacheParameters_l()
void AudioFlinger::DirectOutputThread::flushHw_l()
{
- if (mOutput->stream->flush != NULL) {
- mOutput->stream->flush(mOutput->stream);
- }
+ mOutput->flush();
mHwPaused = false;
}
@@ -4646,7 +4880,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr
size_t audioHALFrames =
(mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
size_t framesWritten =
- mBytesWritten / audio_stream_out_frame_size(mOutput->stream);
+ mBytesWritten / mOutput->getFrameSize();
track->presentationComplete(framesWritten, audioHALFrames);
track->reset();
tracksToRemove->add(track);
@@ -4797,16 +5031,8 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
{
- // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT
- // for delivery downstream as needed. This in-place conversion is safe as
- // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format
- // (AUDIO_FORMAT_PCM_8_BIT is not allowed here).
- if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT,
- mSinkBuffer, mFormat, writeFrames * mChannelCount);
- }
for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames);
+ outputTracks[i]->write(mSinkBuffer, writeFrames);
}
mStandby = false;
return (ssize_t)mSinkBufferSize;
@@ -4833,25 +5059,26 @@ void AudioFlinger::DuplicatingThread::clearOutputTracks()
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
Mutex::Autolock _l(mLock);
- // FIXME explain this formula
- size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
- // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat
- // due to current usage case and restrictions on the AudioBufferProvider.
- // Actual buffer conversion is done in threadLoop_write().
- //
- // TODO: This may change in the future, depending on multichannel
- // (and non int16_t*) support on AF::PlaybackThread::OutputTrack
- OutputTrack *outputTrack = new OutputTrack(thread,
+ // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
+ // Adjust for thread->sampleRate() to determine minimum buffer frame count.
+ // Then triple buffer because Threads do not run synchronously and may not be clock locked.
+ const size_t frameCount =
+ 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
+ // TODO: Consider asynchronous sample rate conversion to handle clock disparity
+ // from different OutputTracks and their associated MixerThreads (e.g. one may
+ // nearly empty and the other may be dropping data).
+
+ sp<OutputTrack> outputTrack = new OutputTrack(thread,
this,
mSampleRate,
- AUDIO_FORMAT_PCM_16_BIT,
+ mFormat,
mChannelMask,
frameCount,
IPCThreadState::self()->getCallingUid());
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
mOutputTracks.add(outputTrack);
- ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+ ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
updateWaitTime_l();
}
}
@@ -4952,8 +5179,8 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
// mFastCaptureNBLogWriter
, mFastTrackAvail(false)
{
- snprintf(mName, kNameLength, "AudioIn_%X", id);
- mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
+ snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id);
+ mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
readInputParameters_l();
@@ -4993,7 +5220,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
}
if (initFastCapture) {
- // create a Pipe for FastMixer to write to, and for us and fast tracks to read from
+ // create a Pipe for FastCapture to write to, and for us and fast tracks to read from
NBAIO_Format format = mInputSource->format();
size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each
size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
@@ -5069,7 +5296,6 @@ failed: ;
// FIXME mNormalSource
}
-
AudioFlinger::RecordThread::~RecordThread()
{
if (mFastCapture != 0) {
@@ -5094,7 +5320,7 @@ AudioFlinger::RecordThread::~RecordThread()
void AudioFlinger::RecordThread::onFirstRef()
{
- run(mName, PRIORITY_URGENT_AUDIO);
+ run(mThreadName, PRIORITY_URGENT_AUDIO);
}
bool AudioFlinger::RecordThread::threadLoop()
@@ -5135,7 +5361,9 @@ reacquire_wakelock:
// sleep with mutex unlocked
if (sleepUs > 0) {
+ ATRACE_BEGIN("sleep");
usleep(sleepUs);
+ ATRACE_END();
sleepUs = 0;
}
@@ -5279,7 +5507,8 @@ reacquire_wakelock:
state->mCommand = FastCaptureState::READ_WRITE;
#if 0 // FIXME
mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
- FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
+ FastThreadDumpState::kSamplingNforLowRamDevice :
+ FastThreadDumpState::kSamplingN);
#endif
didModify = true;
}
@@ -5370,6 +5599,9 @@ reacquire_wakelock:
continue;
}
+ // TODO: This code probably should be moved to RecordTrack.
+ // TODO: Update the activeTrack buffer converter in case of reconfigure.
+
enum {
OVERRUN_UNKNOWN,
OVERRUN_TRUE,
@@ -5384,131 +5616,28 @@ reacquire_wakelock:
size_t framesOut = activeTrack->mSink.frameCount;
LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
- int32_t front = activeTrack->mRsmpInFront;
- ssize_t filled = rear - front;
+ // check available frames and handle overrun conditions
+ // if the record track isn't draining fast enough.
+ bool hasOverrun;
size_t framesIn;
-
- if (filled < 0) {
- // should not happen, but treat like a massive overrun and re-sync
- framesIn = 0;
- activeTrack->mRsmpInFront = rear;
- overrun = OVERRUN_TRUE;
- } else if ((size_t) filled <= mRsmpInFrames) {
- framesIn = (size_t) filled;
- } else {
- // client is not keeping up with server, but give it latest data
- framesIn = mRsmpInFrames;
- activeTrack->mRsmpInFront = front = rear - framesIn;
+ activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
+ if (hasOverrun) {
overrun = OVERRUN_TRUE;
}
-
if (framesOut == 0 || framesIn == 0) {
break;
}
- if (activeTrack->mResampler == NULL) {
- // no resampling
- if (framesIn > framesOut) {
- framesIn = framesOut;
- } else {
- framesOut = framesIn;
- }
- int8_t *dst = activeTrack->mSink.i8;
- while (framesIn > 0) {
- front &= mRsmpInFramesP2 - 1;
- size_t part1 = mRsmpInFramesP2 - front;
- if (part1 > framesIn) {
- part1 = framesIn;
- }
- int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize);
- if (mChannelCount == activeTrack->mChannelCount) {
- memcpy(dst, src, part1 * mFrameSize);
- } else if (mChannelCount == 1) {
- upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src,
- part1);
- } else {
- downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src,
- part1);
- }
- dst += part1 * activeTrack->mFrameSize;
- front += part1;
- framesIn -= part1;
- }
- activeTrack->mRsmpInFront += framesOut;
-
- } else {
- // resampling
- // FIXME framesInNeeded should really be part of resampler API, and should
- // depend on the SRC ratio
- // to keep mRsmpInBuffer full so resampler always has sufficient input
- size_t framesInNeeded;
- // FIXME only re-calculate when it changes, and optimize for common ratios
- // Do not precompute in/out because floating point is not associative
- // e.g. a*b/c != a*(b/c).
- const double in(mSampleRate);
- const double out(activeTrack->mSampleRate);
- framesInNeeded = ceil(framesOut * in / out) + 1;
- ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g",
- framesInNeeded, framesOut, in / out);
- // Although we theoretically have framesIn in circular buffer, some of those are
- // unreleased frames, and thus must be discounted for purpose of budgeting.
- size_t unreleased = activeTrack->mRsmpInUnrel;
- framesIn = framesIn > unreleased ? framesIn - unreleased : 0;
- if (framesIn < framesInNeeded) {
- ALOGV("not enough to resample: have %u frames in but need %u in to "
- "produce %u out given in/out ratio of %.4g",
- framesIn, framesInNeeded, framesOut, in / out);
- size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0;
- LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut);
- if (newFramesOut == 0) {
- break;
- }
- framesInNeeded = ceil(newFramesOut * in / out) + 1;
- ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g",
- framesInNeeded, newFramesOut, out / in);
- LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded);
- ALOGV("success 2: have %u frames in and need %u in to produce %u out "
- "given in/out ratio of %.4g",
- framesIn, framesInNeeded, newFramesOut, in / out);
- framesOut = newFramesOut;
- } else {
- ALOGV("success 1: have %u in and need %u in to produce %u out "
- "given in/out ratio of %.4g",
- framesIn, framesInNeeded, framesOut, in / out);
- }
-
- // reallocate mRsmpOutBuffer as needed; we will grow but never shrink
- if (activeTrack->mRsmpOutFrameCount < framesOut) {
- // FIXME why does each track need it's own mRsmpOutBuffer? can't they share?
- delete[] activeTrack->mRsmpOutBuffer;
- // resampler always outputs stereo
- activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2];
- activeTrack->mRsmpOutFrameCount = framesOut;
- }
-
- // resampler accumulates, but we only have one source track
- memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
- activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut,
- // FIXME how about having activeTrack implement this interface itself?
- activeTrack->mResamplerBufferProvider
- /*this*/ /* AudioBufferProvider* */);
- // ditherAndClamp() works as long as all buffers returned by
- // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
- if (activeTrack->mChannelCount == 1) {
- // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t
- ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer,
- framesOut);
- // the resampler always outputs stereo samples:
- // do post stereo to mono conversion
- downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16,
- (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut);
- } else {
- ditherAndClamp((int32_t *)activeTrack->mSink.raw,
- activeTrack->mRsmpOutBuffer, framesOut);
- }
- // now done with mRsmpOutBuffer
-
- }
+ // Don't allow framesOut to be larger than what is possible with resampling
+ // from framesIn.
+ // This isn't strictly necessary but helps limit buffer resizing in
+ // RecordBufferConverter. TODO: remove when no longer needed.
+ framesOut = min(framesOut,
+ destinationFramesPossible(
+ framesIn, mSampleRate, activeTrack->mSampleRate));
+ // process frames from the RecordThread buffer provider to the RecordTrack buffer
+ framesOut = activeTrack->mRecordBufferConverter->convert(
+ activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);
if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
overrun = OVERRUN_FALSE;
@@ -5649,8 +5778,9 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
// client expresses a preference for FAST, but we get the final say
if (*flags & IAudioFlinger::TRACK_FAST) {
if (
- // use case: callback handler
- (tid != -1) &&
+ // we formerly checked for a callback handler (non-0 tid),
+ // but that is no longer required for TRANSFER_OBTAIN mode
+ //
// frame count is not specified, or is exactly the pipe depth
((frameCount == 0) || (frameCount == mPipeFramesP2)) &&
// PCM data
@@ -5816,12 +5946,9 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac
// was initialized to some value closer to the thread's mRsmpInFront, then the track could
// see previously buffered data before it called start(), but with greater risk of overrun.
- recordTrack->mRsmpInFront = mRsmpInRear;
- recordTrack->mRsmpInUnrel = 0;
- // FIXME why reset?
- if (recordTrack->mResampler != NULL) {
- recordTrack->mResampler->reset();
- }
+ recordTrack->mResamplerBufferProvider->reset();
+ // clear any converter state as new data will be discontinuous
+ recordTrack->mRecordBufferConverter->reset();
recordTrack->mState = TrackBase::STARTING_2;
// signal thread to start
mWaitWorkCV.broadcast();
@@ -5939,15 +6066,17 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a
{
dprintf(fd, "\nInput thread %p:\n", this);
- if (mActiveTracks.size() > 0) {
- dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
- } else {
+ dumpBase(fd, args);
+
+ if (mActiveTracks.size() == 0) {
dprintf(fd, " No active record clients\n");
}
dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
- dumpBase(fd, args);
+ // Make a non-atomic copy of fast capture dump state so it won't change underneath us
+ const FastCaptureDumpState copy(mFastCaptureDumpState);
+ copy.dump(fd);
}
void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
@@ -5995,12 +6124,52 @@ void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args
write(fd, result.string(), result.size());
}
+
+void AudioFlinger::RecordThread::ResamplerBufferProvider::reset()
+{
+ sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
+ RecordThread *recordThread = (RecordThread *) threadBase.get();
+ mRsmpInFront = recordThread->mRsmpInRear;
+ mRsmpInUnrel = 0;
+}
+
+void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
+ size_t *framesAvailable, bool *hasOverrun)
+{
+ sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
+ RecordThread *recordThread = (RecordThread *) threadBase.get();
+ const int32_t rear = recordThread->mRsmpInRear;
+ const int32_t front = mRsmpInFront;
+ const ssize_t filled = rear - front;
+
+ size_t framesIn;
+ bool overrun = false;
+ if (filled < 0) {
+ // should not happen, but treat like a massive overrun and re-sync
+ framesIn = 0;
+ mRsmpInFront = rear;
+ overrun = true;
+ } else if ((size_t) filled <= recordThread->mRsmpInFrames) {
+ framesIn = (size_t) filled;
+ } else {
+ // client is not keeping up with server, but give it latest data
+ framesIn = recordThread->mRsmpInFrames;
+ mRsmpInFront = /* front = */ rear - framesIn;
+ overrun = true;
+ }
+ if (framesAvailable != NULL) {
+ *framesAvailable = framesIn;
+ }
+ if (hasOverrun != NULL) {
+ *hasOverrun = overrun;
+ }
+}
+
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
{
- RecordTrack *activeTrack = mRecordTrack;
- sp<ThreadBase> threadBase = activeTrack->mThread.promote();
+ sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
if (threadBase == 0) {
buffer->frameCount = 0;
buffer->raw = NULL;
@@ -6008,7 +6177,7 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
}
RecordThread *recordThread = (RecordThread *) threadBase.get();
int32_t rear = recordThread->mRsmpInRear;
- int32_t front = activeTrack->mRsmpInFront;
+ int32_t front = mRsmpInFront;
ssize_t filled = rear - front;
// FIXME should not be P2 (don't want to increase latency)
// FIXME if client not keeping up, discard
@@ -6025,17 +6194,16 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
part1 = ask;
}
if (part1 == 0) {
- // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty
- LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved");
+ // out of data is fine since the resampler will return a short-count.
buffer->raw = NULL;
buffer->frameCount = 0;
- activeTrack->mRsmpInUnrel = 0;
+ mRsmpInUnrel = 0;
return NOT_ENOUGH_DATA;
}
buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount;
buffer->frameCount = part1;
- activeTrack->mRsmpInUnrel = part1;
+ mRsmpInUnrel = part1;
return NO_ERROR;
}
@@ -6043,18 +6211,197 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
AudioBufferProvider::Buffer* buffer)
{
- RecordTrack *activeTrack = mRecordTrack;
size_t stepCount = buffer->frameCount;
if (stepCount == 0) {
return;
}
- ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel);
- activeTrack->mRsmpInUnrel -= stepCount;
- activeTrack->mRsmpInFront += stepCount;
+ ALOG_ASSERT(stepCount <= mRsmpInUnrel);
+ mRsmpInUnrel -= stepCount;
+ mRsmpInFront += stepCount;
buffer->raw = NULL;
buffer->frameCount = 0;
}
+AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter(
+ audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+ uint32_t srcSampleRate,
+ audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+ uint32_t dstSampleRate) :
+ mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars
+ // mSrcFormat
+ // mSrcSampleRate
+ // mDstChannelMask
+ // mDstFormat
+ // mDstSampleRate
+ // mSrcChannelCount
+ // mDstChannelCount
+ // mDstFrameSize
+ mBuf(NULL), mBufFrames(0), mBufFrameSize(0),
+ mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0)
+{
+ (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate,
+ dstChannelMask, dstFormat, dstSampleRate);
+}
+
+AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() {
+ free(mBuf);
+ delete mResampler;
+ free(mRsmpOutBuffer);
+}
+
+size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst,
+ AudioBufferProvider *provider, size_t frames)
+{
+ if (mSrcSampleRate == mDstSampleRate) {
+ ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
+ mSrcSampleRate, mSrcFormat, mDstFormat);
+
+ AudioBufferProvider::Buffer buffer;
+ for (size_t i = frames; i > 0; ) {
+ buffer.frameCount = i;
+ status_t status = provider->getNextBuffer(&buffer, 0);
+ if (status != OK || buffer.frameCount == 0) {
+ frames -= i; // cannot fill request.
+ break;
+ }
+ // convert to destination buffer
+ convert(dst, buffer.raw, buffer.frameCount);
+
+ dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
+ i -= buffer.frameCount;
+ provider->releaseBuffer(&buffer);
+ }
+ } else {
+ ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
+ mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat);
+
+ // reallocate mRsmpOutBuffer as needed; we will grow but never shrink
+ if (mRsmpOutFrameCount < frames) {
+ // FIXME why does each track need it's own mRsmpOutBuffer? can't they share?
+ free(mRsmpOutBuffer);
+ // resampler always outputs stereo (FOR NOW)
+ (void)posix_memalign(&mRsmpOutBuffer, 32, frames * FCC_2 * sizeof(int32_t) /*Q4.27*/);
+ mRsmpOutFrameCount = frames;
+ }
+ // resampler accumulates, but we only have one source track
+ memset(mRsmpOutBuffer, 0, frames * FCC_2 * sizeof(int32_t));
+ frames = mResampler->resample((int32_t*)mRsmpOutBuffer, frames, provider);
+
+ // convert to destination buffer
+ convert(dst, mRsmpOutBuffer, frames);
+ }
+ return frames;
+}
+
+status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
+ audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
+ uint32_t srcSampleRate,
+ audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
+ uint32_t dstSampleRate)
+{
+ // quick evaluation if there is any change.
+ if (mSrcFormat == srcFormat
+ && mSrcChannelMask == srcChannelMask
+ && mSrcSampleRate == srcSampleRate
+ && mDstFormat == dstFormat
+ && mDstChannelMask == dstChannelMask
+ && mDstSampleRate == dstSampleRate) {
+ return NO_ERROR;
+ }
+
+ const bool valid =
+ audio_is_input_channel(srcChannelMask)
+ && audio_is_input_channel(dstChannelMask)
+ && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat)
+ && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat)
+ && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX)
+ ; // no upsampling checks for now
+ if (!valid) {
+ return BAD_VALUE;
+ }
+
+ mSrcFormat = srcFormat;
+ mSrcChannelMask = srcChannelMask;
+ mSrcSampleRate = srcSampleRate;
+ mDstFormat = dstFormat;
+ mDstChannelMask = dstChannelMask;
+ mDstSampleRate = dstSampleRate;
+
+ // compute derived parameters
+ mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);
+ mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);
+ mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat);
+
+ // do we need a format buffer?
+ if (mSrcFormat != mDstFormat && mDstChannelCount != mSrcChannelCount) {
+ mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat);
+ } else {
+ mBufFrameSize = 0;
+ }
+ mBufFrames = 0; // force the buffer to be resized.
+
+ // do we need to resample?
+ if (mSrcSampleRate != mDstSampleRate) {
+ if (mResampler != NULL) {
+ delete mResampler;
+ }
+ mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
+ mSrcChannelCount, mDstSampleRate); // may seem confusing...
+ mResampler->setSampleRate(mSrcSampleRate);
+ mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
+ }
+ return NO_ERROR;
+}
+
+void AudioFlinger::RecordThread::RecordBufferConverter::convert(
+ void *dst, /*const*/ void *src, size_t frames)
+{
+ // check if a memcpy will do
+ if (mResampler == NULL
+ && mSrcChannelCount == mDstChannelCount
+ && mSrcFormat == mDstFormat) {
+ memcpy(dst, src,
+ frames * mDstChannelCount * audio_bytes_per_sample(mDstFormat));
+ return;
+ }
+ // reallocate buffer if needed
+ if (mBufFrameSize != 0 && mBufFrames < frames) {
+ free(mBuf);
+ mBufFrames = frames;
+ (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
+ }
+ // do processing
+ if (mResampler != NULL) {
+ // src channel count is always >= 2.
+ void *dstBuf = mBuf != NULL ? mBuf : dst;
+ // ditherAndClamp() works as long as all buffers returned by
+ // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
+ if (mDstChannelCount == 1) {
+ // the resampler always outputs stereo samples.
+ // FIXME: this rewrites back into src
+ ditherAndClamp((int32_t *)src, (const int32_t *)src, frames);
+ downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf,
+ (const int16_t *)src, frames);
+ } else {
+ ditherAndClamp((int32_t *)dstBuf, (const int32_t *)src, frames);
+ }
+ } else if (mSrcChannelCount != mDstChannelCount) {
+ void *dstBuf = mBuf != NULL ? mBuf : dst;
+ if (mSrcChannelCount == 1) {
+ upmix_to_stereo_i16_from_mono_i16((int16_t *)dstBuf, (const int16_t *)src,
+ frames);
+ } else {
+ downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf,
+ (const int16_t *)src, frames);
+ }
+ }
+ if (mSrcFormat != mDstFormat) {
+ void *srcBuf = mBuf != NULL ? mBuf : src;
+ memcpy_by_audio_format(dst, mDstFormat, srcBuf, mSrcFormat,
+ frames * mDstChannelCount);
+ }
+}
+
bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
@@ -6076,7 +6423,7 @@ bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValueP
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
+ if (!audio_is_linear_pcm((audio_format_t) value)) {
status = BAD_VALUE;
} else {
reqFormat = (audio_format_t) value;
@@ -6150,10 +6497,10 @@ bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValueP
}
if (reconfig) {
if (status == BAD_VALUE &&
- reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
- reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
+ audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) &&
+ audio_is_linear_pcm(reqFormat) &&
(mInput->stream->common.get_sample_rate(&mInput->stream->common)
- <= (2 * samplingRate)) &&
+ <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) &&
audio_channel_count_from_in_mask(
mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
(channelMask == AUDIO_CHANNEL_IN_MONO ||
@@ -6224,6 +6571,8 @@ void AudioFlinger::RecordThread::readInputParameters_l()
// The value is somewhat arbitrary, and could probably be even larger.
// A larger value should allow more old data to be read after a track calls start(),
// without increasing latency.
+ //
+ // Note this is independent of the maximum downsampling ratio permitted for capture.
mRsmpInFrames = mFrameCount * 7;
mRsmpInFramesP2 = roundup(mRsmpInFrames);
delete[] mRsmpInBuffer;
@@ -6412,4 +6761,4 @@ void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *co
config->ext.mix.usecase.source = mAudioSource;
}
-}; // namespace android
+} // namespace android
generated by cgit v1.1 at 2024-06-07 12:33:01 +0000