Update the getNextBuffer signatures to have an additional (but
ignorable) parameter used to implement timed audio output.
Change-Id: Icb98e4af4d136c234ae8c5cbb4a5de4350b436bb
return OK;
}
-status_t VideoEditorSRC::getNextBuffer(AudioBufferProvider::Buffer *pBuffer) {
+status_t VideoEditorSRC::getNextBuffer(AudioBufferProvider::Buffer *pBuffer, int64_t pts) {
ALOGV("getNextBuffer %d, chan = %d", pBuffer->frameCount, mChannelCnt);
uint32_t done = 0;
uint32_t want = pBuffer->frameCount * mChannelCnt * 2;
virtual status_t read (
MediaBuffer **buffer, const ReadOptions *options = NULL);
- virtual status_t getNextBuffer(Buffer* buffer);
+ virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
virtual void releaseBuffer(Buffer* buffer);
// Sampling freqencies
public:
- virtual status_t getNextBuffer(Buffer* buffer);
+ virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
virtual void releaseBuffer(Buffer* buffer);
enum { //Sampling freq
#define MAX_SAMPLEDURATION_FOR_CONVERTION 40 //ms
-status_t VideoEditorResampler::getNextBuffer(AudioBufferProvider::Buffer *pBuffer) {
+status_t VideoEditorResampler::getNextBuffer(AudioBufferProvider::Buffer *pBuffer, int64_t pts) {
uint32_t dataSize = pBuffer->frameCount * this->nbChannels * sizeof(int16_t);
mTmpInBuffer = (int16_t*)malloc(dataSize);