Multi-client recording continued
Remove mReqChannelCount and mReqSampleRate.
Increase number of input frames needed for sample rate converter.
Add TODO comments to investigate later.
Fix bugs in sample rate conversion.
Change-Id: I3aa54b862843d518b6e4ece9a7f52fd27d741629
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 43613aa..33c1178 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -4451,8 +4451,6 @@
AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
- uint32_t sampleRate,
- audio_channel_mask_t channelMask,
audio_io_handle_t id,
audio_devices_t outDevice,
audio_devices_t inDevice
@@ -4463,10 +4461,7 @@
ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL),
// mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l()
- mRsmpInRear(0),
- // FIXME these should be per-track, so this is only the initial track?
- mReqChannelCount(popcount(channelMask)),
- mReqSampleRate(sampleRate)
+ mRsmpInRear(0)
#ifdef TEE_SINK
, mTeeSink(teeSink)
#endif
@@ -4598,13 +4593,6 @@
case TrackBase::STARTING_2:
doBroadcast = true;
- if (mReqChannelCount != activeTrack->channelCount()) {
- ALOGW("wrong channel count");
- mActiveTracks.remove(activeTrack);
- mActiveTracksGen++;
- size--;
- continue;
- }
mStandby = false;
activeTrack->mState = TrackBase::ACTIVE;
break;
@@ -4715,6 +4703,10 @@
overrun = OVERRUN_TRUE;
}
+ if (framesOut == 0 || framesIn == 0) {
+ break;
+ }
+
if (activeTrack->mResampler == NULL) {
// no resampling
if (framesIn > framesOut) {
@@ -4730,7 +4722,7 @@
part1 = framesIn;
}
int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize);
- if (mChannelCount == mReqChannelCount) {
+ if (mChannelCount == activeTrack->mChannelCount) {
memcpy(dst, src, part1 * mFrameSize);
} else if (mChannelCount == 1) {
upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src,
@@ -4754,14 +4746,29 @@
// FIXME only re-calculate when it changes, and optimize for common ratios
double inOverOut = (double) mSampleRate / activeTrack->mSampleRate;
double outOverIn = (double) activeTrack->mSampleRate / mSampleRate;
- framesInNeeded = framesOut * inOverOut;
+ framesInNeeded = ceil(framesOut * inOverOut) + 1;
if (framesIn < framesInNeeded) {
- ALOGV("not enough to resample: have %u but need %u to produce %u",
- framesIn, framesInNeeded, framesOut);
- size_t newFramesOut = framesIn * outOverIn;
- size_t newFramesInNeeded = newFramesOut * inOverOut;
- LOG_ALWAYS_FATAL_IF(framesIn < newFramesInNeeded);
- framesOut = newFramesOut;
+ ALOGV("not enough to resample: have %u but need %u to produce %u "
+ "given in/out ratio of %.4g",
+ framesIn, framesInNeeded, framesOut, inOverOut);
+ size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0;
+ size_t newFramesInNeeded = ceil(newFramesOut * inOverOut) + 1;
+ ALOGV("now need %u frames to produce %u given out/in ratio of %.4g",
+ newFramesInNeeded, newFramesOut, outOverIn);
+ if (framesIn < newFramesInNeeded) {
+ ALOGE("failure: have %u but need %u", framesIn, newFramesInNeeded);
+ framesOut = 0;
+ } else {
+ ALOGV("success 2: have %u and need %u to produce %u "
+ "given in/out ratio of %.4g",
+ framesIn, newFramesInNeeded, newFramesOut, inOverOut);
+ LOG_ALWAYS_FATAL_IF(newFramesOut > framesOut);
+ framesOut = newFramesOut;
+ }
+ } else {
+ ALOGI("success 1: have %u and need %u to produce %u "
+ "given in/out ratio of %.4g",
+ framesIn, framesInNeeded, framesOut, inOverOut);
}
// reallocate mRsmpOutBuffer as needed; we will grow but never shrink
@@ -4779,7 +4786,7 @@
/*this*/ /* AudioBufferProvider* */);
// ditherAndClamp() works as long as all buffers returned by
// activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
- if (mReqChannelCount == 1) {
+ if (activeTrack->mChannelCount == 1) {
// temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer,
framesOut);
@@ -4826,9 +4833,6 @@
}
if (framesOut == 0) {
- if (overrun == OVERRUN_UNKNOWN) {
- overrun = OVERRUN_TRUE;
- }
break;
}
}
@@ -4849,7 +4853,6 @@
activeTrack->clearOverflow();
break;
case OVERRUN_UNKNOWN:
- LOG_FATAL("OVERRUN_UNKNOWN");
break;
}
@@ -5025,7 +5028,7 @@
} else {
// do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
recordTrack->mFramesToDrop = -
- ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000);
+ ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000);
}
}
@@ -5042,6 +5045,9 @@
return status;
}
+ // TODO consider other ways of handling this, such as changing the state to :STARTING and
+ // adding the track to mActiveTracks after returning from AudioSystem::startInput(),
+ // or using a separate command thread
recordTrack->mState = TrackBase::STARTING_1;
mActiveTracks.add(recordTrack);
mActiveTracksGen++;
@@ -5176,8 +5182,6 @@
if (mActiveTracks.size() > 0) {
fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
- fdprintf(fd, " Out channel count: %u\n", mReqChannelCount);
- fdprintf(fd, " Out sample rate: %u\n", mReqSampleRate);
} else {
fdprintf(fd, " No active record clients\n");
}
@@ -5260,7 +5264,7 @@
}
if (part1 == 0) {
// Higher-level should keep mRsmpInBuffer full, and not call resampler if empty
- ALOGE("RecordThread::getNextBuffer() starved");
+ LOG_FATAL("RecordThread::getNextBuffer() starved");
buffer->raw = NULL;
buffer->frameCount = 0;
activeTrack->mRsmpInUnrel = 0;
@@ -5299,11 +5303,14 @@
AudioParameter param = AudioParameter(keyValuePair);
int value;
audio_format_t reqFormat = mFormat;
- uint32_t reqSamplingRate = mReqSampleRate;
- audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount);
+ uint32_t samplingRate = mSampleRate;
+ audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount);
+ // TODO Investigate when this code runs. Check with audio policy when a sample rate and
+ // channel count change can be requested. Do we mandate the first client defines the
+ // HAL sampling rate and channel count or do we allow changes on the fly?
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
- reqSamplingRate = value;
+ samplingRate = value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
@@ -5319,7 +5326,7 @@
if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) {
status = BAD_VALUE;
} else {
- reqChannelMask = mask;
+ channelMask = mask;
reconfig = true;
}
}
@@ -5384,11 +5391,11 @@
reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
(mInput->stream->common.get_sample_rate(&mInput->stream->common)
- <= (2 * reqSamplingRate)) &&
+ <= (2 * samplingRate)) &&
popcount(mInput->stream->common.get_channels(&mInput->stream->common))
<= FCC_2 &&
- (reqChannelMask == AUDIO_CHANNEL_IN_MONO ||
- reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) {
+ (channelMask == AUDIO_CHANNEL_IN_MONO ||
+ channelMask == AUDIO_CHANNEL_IN_STEREO)) {
status = NO_ERROR;
}
if (status == NO_ERROR) {
@@ -5468,8 +5475,8 @@
// Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount];
- // mReqSampleRate and mReqChannelCount are constant due to AudioRecord API constraints.
- // But if mSampleRate or mChannelCount changes, how will that affect active tracks?
+ // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
+ // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
}
uint32_t AudioFlinger::RecordThread::getInputFramesLost()