Merge "Allow multichannel FLAC files"
diff --git a/media/libstagefright/FLACExtractor.cpp b/media/libstagefright/FLACExtractor.cpp
index 29bb056..098fcf9 100644
--- a/media/libstagefright/FLACExtractor.cpp
+++ b/media/libstagefright/FLACExtractor.cpp
@@ -122,7 +122,7 @@
// media buffers
size_t mMaxBufferSize;
MediaBufferGroup *mGroup;
- void (*mCopy)(short *dst, const int *const *src, unsigned nSamples);
+ void (*mCopy)(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels);
// handle to underlying libFLAC parser
FLAC__StreamDecoder *mDecoder;
@@ -380,14 +380,14 @@
// Copy samples from FLAC native 32-bit non-interleaved to 16-bit interleaved.
// These are candidates for optimization if needed.
-static void copyMono8(short *dst, const int *const *src, unsigned nSamples)
+static void copyMono8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
{
for (unsigned i = 0; i < nSamples; ++i) {
*dst++ = src[0][i] << 8;
}
}
-static void copyStereo8(short *dst, const int *const *src, unsigned nSamples)
+static void copyStereo8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
{
for (unsigned i = 0; i < nSamples; ++i) {
*dst++ = src[0][i] << 8;
@@ -395,14 +395,23 @@
}
}
-static void copyMono16(short *dst, const int *const *src, unsigned nSamples)
+static void copyMultiCh8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
+{
+ for (unsigned i = 0; i < nSamples; ++i) {
+ for (unsigned c = 0; c < nChannels; ++c) {
+ *dst++ = src[c][i] << 8;
+ }
+ }
+}
+
+static void copyMono16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
{
for (unsigned i = 0; i < nSamples; ++i) {
*dst++ = src[0][i];
}
}
-static void copyStereo16(short *dst, const int *const *src, unsigned nSamples)
+static void copyStereo16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
{
for (unsigned i = 0; i < nSamples; ++i) {
*dst++ = src[0][i];
@@ -410,16 +419,25 @@
}
}
+static void copyMultiCh16(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
+{
+ for (unsigned i = 0; i < nSamples; ++i) {
+ for (unsigned c = 0; c < nChannels; ++c) {
+ *dst++ = src[c][i];
+ }
+ }
+}
+
// 24-bit versions should do dithering or noise-shaping, here or in AudioFlinger
-static void copyMono24(short *dst, const int *const *src, unsigned nSamples)
+static void copyMono24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
{
for (unsigned i = 0; i < nSamples; ++i) {
*dst++ = src[0][i] >> 8;
}
}
-static void copyStereo24(short *dst, const int *const *src, unsigned nSamples)
+static void copyStereo24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
{
for (unsigned i = 0; i < nSamples; ++i) {
*dst++ = src[0][i] >> 8;
@@ -427,7 +445,16 @@
}
}
-static void copyTrespass(short *dst, const int *const *src, unsigned nSamples)
+static void copyMultiCh24(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
+{
+ for (unsigned i = 0; i < nSamples; ++i) {
+ for (unsigned c = 0; c < nChannels; ++c) {
+ *dst++ = src[c][i] >> 8;
+ }
+ }
+}
+
+static void copyTrespass(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
{
TRESPASS();
}
@@ -507,11 +534,7 @@
}
if (mStreamInfoValid) {
// check channel count
- switch (getChannels()) {
- case 1:
- case 2:
- break;
- default:
+ if (getChannels() == 0 || getChannels() > 8) {
ALOGE("unsupported channel count %u", getChannels());
return NO_INIT;
}
@@ -536,9 +559,10 @@
case 32000:
case 44100:
case 48000:
+ case 88200:
+ case 96000:
break;
default:
- // 96000 would require a proper downsampler in AudioFlinger
ALOGE("unsupported sample rate %u", getSampleRate());
return NO_INIT;
}
@@ -546,17 +570,20 @@
static const struct {
unsigned mChannels;
unsigned mBitsPerSample;
- void (*mCopy)(short *dst, const int *const *src, unsigned nSamples);
+ void (*mCopy)(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels);
} table[] = {
{ 1, 8, copyMono8 },
{ 2, 8, copyStereo8 },
+ { 8, 8, copyMultiCh8 },
{ 1, 16, copyMono16 },
{ 2, 16, copyStereo16 },
+ { 8, 16, copyMultiCh16 },
{ 1, 24, copyMono24 },
{ 2, 24, copyStereo24 },
+ { 8, 24, copyMultiCh24 },
};
for (unsigned i = 0; i < sizeof(table)/sizeof(table[0]); ++i) {
- if (table[i].mChannels == getChannels() &&
+ if (table[i].mChannels >= getChannels() &&
table[i].mBitsPerSample == getBitsPerSample()) {
mCopy = table[i].mCopy;
break;
@@ -640,7 +667,7 @@
short *data = (short *) buffer->data();
buffer->set_range(0, bufferSize);
// copy PCM from FLAC write buffer to our media buffer, with interleaving
- (*mCopy)(data, mWriteBuffer, blocksize);
+ (*mCopy)(data, mWriteBuffer, blocksize, getChannels());
// fill in buffer metadata
CHECK(mWriteHeader.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER);
FLAC__uint64 sampleNumber = mWriteHeader.number.sample_number;
