AudioRecord and HAL input stream must be 16-bit PCM only
Currently there are 16-bit PCM assumptions in several places for capture:
- resampler API
- mRsmpInBuffer and mRsmpOutBuffer
- RecordThread::threadLoop upmix, downmix, and resampling
- possibly other places
Until those assumptions are removed, this CL enforces 16-bit PCM in both
client and server at all places where a format is checked.
Change-Id: I08b0570bff626ad0d341804825a72c14e61b4233
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index f27d908..c3a922a 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -3737,8 +3737,7 @@
framesIn = framesOut;
mRsmpInIndex += framesIn;
framesOut -= framesIn;
- if (mChannelCount == mReqChannelCount ||
- mFormat != AUDIO_FORMAT_PCM_16_BIT) {
+ if (mChannelCount == mReqChannelCount) {
memcpy(dst, src, framesIn * mFrameSize);
} else {
if (mChannelCount == 1) {
@@ -3752,9 +3751,7 @@
}
if (framesOut && mFrameCount == mRsmpInIndex) {
void *readInto;
- if (framesOut == mFrameCount &&
- (mChannelCount == mReqChannelCount ||
- mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
+ if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) {
readInto = buffer.raw;
framesOut = 0;
} else {
@@ -4224,8 +4221,12 @@
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- reqFormat = (audio_format_t) value;
- reconfig = true;
+ if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
+ status = BAD_VALUE;
+ } else {
+ reqFormat = (audio_format_t) value;
+ reconfig = true;
+ }
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
reqChannelCount = popcount(value);
@@ -4366,6 +4367,9 @@
mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
+ if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
+ ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
+ }
mFrameSize = audio_stream_frame_size(&mInput->stream->common);
mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
mFrameCount = mInputBytes / mFrameSize;