diff options
Diffstat (limited to 'media/libmedia/AudioRecord.cpp')
-rw-r--r-- | media/libmedia/AudioRecord.cpp | 53 |
1 files changed, 9 insertions, 44 deletions
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; |