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-rw-r--r--include/media/AudioRecord.h1
-rw-r--r--media/libmedia/AudioRecord.cpp53
-rw-r--r--media/libmedia/IAudioFlinger.cpp3
-rwxr-xr-xservices/audioflinger/Threads.cpp66
4 files changed, 43 insertions, 80 deletions
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index f9c7efd..4edc1bf 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -461,6 +461,7 @@ private:
// for notification APIs
uint32_t mNotificationFramesReq; // requested number of frames between each
// notification callback
+ // as specified in constructor or set()
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback
bool mRefreshRemaining; // processAudioBuffer() should refresh
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index c474a25..80c8c5e 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -211,7 +211,7 @@ status_t AudioRecord::set(
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
- mNotificationFramesAct = 0;
+ // mNotificationFramesAct is initialized in openRecord_l
if (sessionId == AUDIO_SESSION_ALLOCATE) {
mSessionId = AudioSystem::newAudioSessionId();
@@ -444,42 +444,6 @@ status_t AudioRecord::openRecord_l(size_t epoch)
}
}
- // FIXME Assume double buffering, because we don't know the true HAL sample rate
- const uint32_t nBuffering = 2;
-
- mNotificationFramesAct = mNotificationFramesReq;
- size_t frameCount = mReqFrameCount;
-
- if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
- // validate framecount
- // If fast track was not requested, this preserves
- // the old behavior of validating on client side.
- // FIXME Eventually the validation should be done on server side
- // regardless of whether it's a fast or normal track. It's debatable
- // whether to account for the input latency to provision buffers appropriately.
- size_t minFrameCount;
- status = AudioRecord::getMinFrameCount(&minFrameCount,
- mSampleRate, mFormat, mChannelMask);
- if (status != NO_ERROR) {
- ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; "
- "status %d",
- mSampleRate, mFormat, mChannelMask, status);
- return status;
- }
-
- if (frameCount == 0) {
- frameCount = minFrameCount;
- } else if (frameCount < minFrameCount) {
- ALOGE("frameCount %zu < minFrameCount %zu", frameCount, minFrameCount);
- return BAD_VALUE;
- }
-
- // Make sure that application is notified with sufficient margin before overrun
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
- mNotificationFramesAct = frameCount/2;
- }
- }
-
audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat,
mChannelMask, mSessionId, mFlags);
if (input == AUDIO_IO_HANDLE_NONE) {
@@ -492,12 +456,13 @@ status_t AudioRecord::openRecord_l(size_t epoch)
// Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
// we must release it ourselves if anything goes wrong.
+ size_t frameCount = mReqFrameCount;
size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
// but we will still need the original value also
int originalSessionId = mSessionId;
// The notification frame count is the period between callbacks, as suggested by the server.
- size_t notificationFrames;
+ size_t notificationFrames = mNotificationFramesReq;
sp<IMemory> iMem; // for cblk
sp<IMemory> bufferMem;
@@ -576,14 +541,14 @@ status_t AudioRecord::openRecord_l(size_t epoch)
// once denied, do not request again if IAudioRecord is re-created
mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
}
- // Theoretically double-buffering is not required for fast tracks,
- // due to tighter scheduling. But in practice, to accomodate kernels with
- // scheduling jitter, and apps with computation jitter, we use double-buffering.
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
- mNotificationFramesAct = frameCount/nBuffering;
- }
}
+ // Make sure that application is notified with sufficient margin before overrun
+ if (notificationFrames == 0 || notificationFrames > frameCount) {
+ ALOGW("Received notificationFrames %zu for frameCount %zu", notificationFrames, frameCount);
+ }
+ mNotificationFramesAct = notificationFrames;
+
// We retain a copy of the I/O handle, but don't own the reference
mInput = input;
mRefreshRemaining = true;
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 7795fdb..bd7ea46 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -197,6 +197,7 @@ public:
lSessionId = *sessionId;
}
data.writeInt32(lSessionId);
+ data.writeInt64(notificationFrames != NULL ? *notificationFrames : 0);
cblk.clear();
buffers.clear();
status_t lStatus = remote()->transact(OPEN_RECORD, data, &reply);
@@ -966,7 +967,7 @@ status_t BnAudioFlinger::onTransact(
track_flags_t flags = (track_flags_t) data.readInt32();
pid_t tid = (pid_t) data.readInt32();
int sessionId = data.readInt32();
- size_t notificationFrames = 0;
+ size_t notificationFrames = data.readInt64();
sp<IMemory> cblk;
sp<IMemory> buffers;
status_t status;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index e17aa98..0f555c0 100755
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5458,21 +5458,14 @@ 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 and frame count is default or at least as large as HAL
- (
- (tid != -1) &&
- ((frameCount == 0) /*||
- // FIXME must be equal to pipe depth, so don't allow it to be specified by client
- // FIXME not necessarily true, should be native frame count for native SR!
- (frameCount >= mFrameCount)*/)
- ) &&
+ // use case: callback handler
+ (tid != -1) &&
+ // frame count is not specified, or is exactly the pipe depth
+ ((frameCount == 0) || (frameCount == mPipeFramesP2)) &&
// PCM data
audio_is_linear_pcm(format) &&
// native format
(format == mFormat) &&
- // mono or stereo
- ( (channelMask == AUDIO_CHANNEL_IN_MONO) ||
- (channelMask == AUDIO_CHANNEL_IN_STEREO) ) &&
// native channel mask
(channelMask == mChannelMask) &&
// native hardware sample rate
@@ -5482,40 +5475,43 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
// there are sufficient fast track slots available
mFastTrackAvail
) {
- // if frameCount not specified, then it defaults to pipe frame count
- if (frameCount == 0) {
- frameCount = mPipeFramesP2;
- }
- ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
+ ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u",
frameCount, mFrameCount);
} else {
- ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d "
- "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
+ ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u "
+ "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
"hasFastCapture=%d tid=%d mFastTrackAvail=%d",
- frameCount, mFrameCount, format,
- audio_is_linear_pcm(format),
- channelMask, sampleRate, mSampleRate, hasFastCapture(), tid, mFastTrackAvail);
+ frameCount, mFrameCount, mPipeFramesP2,
+ format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate,
+ hasFastCapture(), tid, mFastTrackAvail);
*flags &= ~IAudioFlinger::TRACK_FAST;
- // FIXME It's not clear that we need to enforce this any more, since we have a pipe.
- // For compatibility with AudioRecord calculation, buffer depth is forced
- // to be at least 2 x the record thread 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.
- // FIXME It's not clear how input latency actually matters. Perhaps this should be 0.
- uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream);
- size_t mNormalFrameCount = 2048; // FIXME
- uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
- if (minBufCount < 2) {
- minBufCount = 2;
+ }
+ }
+
+ // compute track buffer size in frames, and suggest the notification frame count
+ if (*flags & IAudioFlinger::TRACK_FAST) {
+ // fast track: frame count is exactly the pipe depth
+ frameCount = mPipeFramesP2;
+ // ignore requested notificationFrames, and always notify exactly once every HAL buffer
+ *notificationFrames = mFrameCount;
+ } else {
+ // not fast track: frame count is at least 2 HAL buffers and at least 20 ms
+ size_t minFrameCount = ((int64_t) mFrameCount * 2 * sampleRate + mSampleRate - 1) /
+ mSampleRate;
+ if (frameCount < minFrameCount) {
+ frameCount = minFrameCount;
}
- size_t minFrameCount = mNormalFrameCount * minBufCount;
+ minFrameCount = (sampleRate * 20 / 1000 + 1) & ~1;
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
- }
+ // notification is forced to be at least double-buffering
+ size_t maxNotification = frameCount / 2;
+ if (*notificationFrames == 0 || *notificationFrames > maxNotification) {
+ *notificationFrames = maxNotification;
+ }
}
*pFrameCount = frameCount;
- *notificationFrames = 0; // FIXME implement
lStatus = initCheck();
if (lStatus != NO_ERROR) {
generated by cgit v1.1 at 2024-05-03 11:31:35 +0000