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ffmpeg encoder: do not use deprecatred function

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Replace avcodec_encode_audio with avcodec_encode_audio2. The latter
provides us with more information on the encoded data, so we can avoid
guessing things on our own. It also handles memory allocations on its
own, which fix some cases where we would provide a too small buffer.
This commit is contained in:
Adrien Destugues 2015-08-29 23:08:37 +02:00
parent 75d1eb3a59
commit 0717601a7f
1 changed files with 64 additions and 29 deletions
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93
src/add-ons/media/plugins/ffmpeg/AVCodecEncoder.cpp
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@ -591,43 +591,78 @@ status_t
AVCodecEncoder::_EncodeAudio(const uint8* buffer, size_t bufferSize,
int64 frameCount, media_encode_info* info)
{
// Encode one audio chunk/frame. The bufferSize has already been adapted
// to the needed size for fContext->frame_size, or we are writing raw
// audio.
int usedBytes = avcodec_encode_audio(fContext, fChunkBuffer,
bufferSize, reinterpret_cast<const short*>(buffer));
status_t ret;
if (usedBytes < 0) {
TRACE(" avcodec_encode_audio() failed: %d\n", usedBytes);
return B_ERROR;
// Encode one audio chunk/frame.
AVPacket packet;
av_init_packet(&packet);
// By leaving these NULL, we let the encoder allocate memory as it needs.
// This way we don't risk iving a too small buffer.
packet.data = NULL;
packet.size = 0;
// We need to wrap our input data into an AVFrame structure.
AVFrame frame;
int gotPacket = 0;
if (buffer) {
avcodec_get_frame_defaults(&frame);
frame.nb_samples = frameCount;
ret = avcodec_fill_audio_frame(&frame, fContext->channels,
fContext->sample_fmt, (const uint8_t *) buffer, bufferSize, 1);
if (ret != 0)
return B_ERROR;
/* Set the presentation time of the frame */
frame.pts = (bigtime_t)(fFramesWritten * 1000000LL
/ fInputFormat.u.raw_audio.frame_rate);
fFramesWritten += frame.nb_samples;
ret = avcodec_encode_audio2(fContext, &packet, &frame, &gotPacket);
} else {
// If called with NULL, ask the encoder to flush any buffers it may
// have pending.
ret = avcodec_encode_audio2(fContext, &packet, NULL, &gotPacket);
}
if (usedBytes == 0)
return B_OK;
// // Maybe we need to use this PTS to calculate start_time:
// if (fContext->coded_frame->pts != kNoPTSValue) {
// TRACE(" codec frame PTS: %lld (codec time_base: %d/%d)\n",
// fContext->coded_frame->pts, fContext->time_base.num,
// fContext->time_base.den);
// } else {
// TRACE(" codec frame PTS: N/A (codec time_base: %d/%d)\n",
// fContext->time_base.num, fContext->time_base.den);
// }
if (buffer && frame.extended_data != frame.data)
av_freep(&frame.extended_data);
// Setup media_encode_info, most important is the time stamp.
info->start_time = (bigtime_t)(fFramesWritten * 1000000LL
/ fInputFormat.u.raw_audio.frame_rate);
info->flags = B_MEDIA_KEY_FRAME;
// Write the chunk
status_t ret = WriteChunk(fChunkBuffer, usedBytes, info);
if (ret != B_OK) {
TRACE(" error writing chunk: %s\n", strerror(ret));
return ret;
if (ret != 0) {
TRACE(" avcodec_encode_audio() failed: %ld\n", ret);
return B_ERROR;
}
fFramesWritten += frameCount;
if (gotPacket) {
if (fContext->coded_frame) {
// Store information about the coded frame in the context.
fContext->coded_frame->pts = packet.pts;
fContext->coded_frame->key_frame = !!(packet.flags & AV_PKT_FLAG_KEY);
}
// Setup media_encode_info, most important is the time stamp.
info->start_time = packet.pts;
if (packet.flags & AV_PKT_FLAG_KEY)
info->flags = B_MEDIA_KEY_FRAME;
else
info->flags = 0;
// We got a packet out of the encoder, write it to the output stream
ret = WriteChunk(packet.data, packet.size, info);
if (ret != B_OK) {
TRACE(" error writing chunk: %s\n", strerror(ret));
av_free_packet(&packet);
return ret;
}
}
av_free_packet(&packet);
return B_OK;
}