Add RecordBufferConverter for RecordThread data processing
Change-Id: Ia3aab8590cd41e8a7cba0a7345d70d2866d92045
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 204a9d6..5537ca2 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -34,6 +34,7 @@
IAudioFlinger::track_flags_t flags,
track_type type);
virtual ~RecordTrack();
+ virtual status_t initCheck() const;
virtual status_t start(AudioSystem::sync_event_t event, int triggerSession);
virtual void stop();
@@ -66,13 +67,6 @@
bool mOverflow; // overflow on most recent attempt to fill client buffer
- // updated by RecordThread::readInputParameters_l()
- AudioResampler *mResampler;
-
- // interleaved stereo pairs of fixed-point Q4.27
- int32_t *mRsmpOutBuffer;
- // current allocated frame count for the above, which may be larger than needed
- size_t mRsmpOutFrameCount;
size_t mRsmpInUnrel; // unreleased frames remaining from
// most recent getNextBuffer
@@ -93,7 +87,10 @@
ssize_t mFramesToDrop;
// used by resampler to find source frames
- ResamplerBufferProvider *mResamplerBufferProvider;
+ ResamplerBufferProvider *mResamplerBufferProvider;
+
+ // used by the record thread to convert frames to proper destination format
+ RecordBufferConverter *mRecordBufferConverter;
};
// playback track, used by PatchPanel
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 4efb3d7..7e71613 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5290,7 +5290,6 @@
// FIXME mNormalSource
}
-
AudioFlinger::RecordThread::~RecordThread()
{
if (mFastCapture != 0) {
@@ -5594,6 +5593,8 @@
continue;
}
+ // TODO: Update the activeTrack buffer converter in case of reconfigure.
+
enum {
OVERRUN_UNKNOWN,
OVERRUN_TRUE,
@@ -5630,109 +5631,9 @@
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
-
- }
+ // 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;
@@ -6043,10 +5944,8 @@
recordTrack->mRsmpInFront = mRsmpInRear;
recordTrack->mRsmpInUnrel = 0;
- // FIXME why reset?
- if (recordTrack->mResampler != NULL) {
- recordTrack->mResampler->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();
@@ -6282,6 +6181,186 @@
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)
{
@@ -6303,7 +6382,7 @@
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;
@@ -6377,10 +6456,10 @@
}
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 ||
@@ -6451,6 +6530,8 @@
// 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;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index d600ea9..053d2e7 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1049,6 +1049,87 @@
RecordTrack * const mRecordTrack;
};
+ /* The RecordBufferConverter is used for format, channel, and sample rate
+ * conversion for a RecordTrack.
+ *
+ * TODO: Self contained, so move to a separate file later.
+ *
+ * RecordBufferConverter uses the convert() method rather than exposing a
+ * buffer provider interface; this is to save a memory copy.
+ */
+ class RecordBufferConverter
+ {
+ public:
+ 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);
+
+ ~RecordBufferConverter();
+
+ /* Converts input data from an AudioBufferProvider by format, channelMask,
+ * and sampleRate to a destination buffer.
+ *
+ * Parameters
+ * dst: buffer to place the converted data.
+ * provider: buffer provider to obtain source data.
+ * frames: number of frames to convert
+ *
+ * Returns the number of frames converted.
+ */
+ size_t convert(void *dst, AudioBufferProvider *provider, size_t frames);
+
+ // returns NO_ERROR if constructor was successful
+ status_t initCheck() const {
+ // mSrcChannelMask set on successful updateParameters
+ return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT;
+ }
+
+ // allows dynamic reconfigure of all parameters
+ status_t 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);
+
+ // called to reset resampler buffers on record track discontinuity
+ void reset() {
+ if (mResampler != NULL) {
+ mResampler->reset();
+ }
+ }
+
+ private:
+ // internal convert function for format and channel mask.
+ void convert(void *dst, /*const*/ void *src, size_t frames);
+
+ // user provided information
+ audio_channel_mask_t mSrcChannelMask;
+ audio_format_t mSrcFormat;
+ uint32_t mSrcSampleRate;
+ audio_channel_mask_t mDstChannelMask;
+ audio_format_t mDstFormat;
+ uint32_t mDstSampleRate;
+
+ // derived information
+ uint32_t mSrcChannelCount;
+ uint32_t mDstChannelCount;
+ size_t mDstFrameSize;
+
+ // format conversion buffer
+ void *mBuf;
+ size_t mBufFrames;
+ size_t mBufFrameSize;
+
+ // resampler info
+ AudioResampler *mResampler;
+ // interleaved stereo pairs of fixed-point Q4.27 or float depending on resampler
+ void *mRsmpOutBuffer;
+ // current allocated frame count for the above, which may be larger than needed
+ size_t mRsmpOutFrameCount;
+ };
+
#include "RecordTracks.h"
RecordThread(const sp<AudioFlinger>& audioFlinger,
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index dc9f249..b2f44f2 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1989,7 +1989,7 @@
((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
type),
- mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
+ mOverflow(false),
// See real initialization of mRsmpInFront at RecordThread::start()
mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
{
@@ -1997,21 +1997,23 @@
return;
}
+ mRecordBufferConverter = new RecordBufferConverter(
+ thread->mChannelMask, thread->mFormat, thread->mSampleRate,
+ channelMask, format, sampleRate);
+ // Check if the RecordBufferConverter construction was successful.
+ // If not, don't continue with construction.
+ //
+ // NOTE: It would be extremely rare that the record track cannot be created
+ // for the current device, but a pending or future device change would make
+ // the record track configuration valid.
+ if (mRecordBufferConverter->initCheck() != NO_ERROR) {
+ ALOGE("RecordTrack unable to create record buffer converter");
+ return;
+ }
+
mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack());
-
- uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
- // FIXME I don't understand either of the channel count checks
- if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
- channelCount <= FCC_2) {
- // sink SR
- mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
- thread->mChannelCount, sampleRate);
- // source SR
- mResampler->setSampleRate(thread->mSampleRate);
- mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
- mResamplerBufferProvider = new ResamplerBufferProvider(this);
- }
+ mResamplerBufferProvider = new ResamplerBufferProvider(this);
if (flags & IAudioFlinger::TRACK_FAST) {
ALOG_ASSERT(thread->mFastTrackAvail);
@@ -2022,11 +2024,19 @@
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
ALOGV("%s", __func__);
- delete mResampler;
- delete[] mRsmpOutBuffer;
+ delete mRecordBufferConverter;
delete mResamplerBufferProvider;
}
+status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const
+{
+ status_t status = TrackBase::initCheck();
+ if (status == NO_ERROR && mServerProxy == 0) {
+ status = BAD_VALUE;
+ }
+ return status;
+}
+
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts __unused)