This reverts commit 14b654326d.
Unfortunately that changeset causes a regression on GCC 2, which
makes playback of (some?) video impossible. This is due to Libavcodec
being miscompiled, which requires gcc >= 4.2
Resolves the regression of #8856, but does not fix the root issue.
705 lines
19 KiB
C++
705 lines
19 KiB
C++
/*
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* Copyright 2009-2010, Stephan Amßus <[email protected]>
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* All rights reserved. Distributed under the terms of the MIT license.
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*/
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#include "AVCodecEncoder.h"
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#include <new>
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#include <stdio.h>
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#include <string.h>
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#include <Application.h>
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#include <Roster.h>
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extern "C" {
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#include "rational.h"
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}
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#include "EncoderTable.h"
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#include "gfx_util.h"
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#undef TRACE
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//#define TRACE_AV_CODEC_ENCODER
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#ifdef TRACE_AV_CODEC_ENCODER
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# define TRACE printf
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# define TRACE_IO(a...)
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#else
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# define TRACE(a...)
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# define TRACE_IO(a...)
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#endif
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static const size_t kDefaultChunkBufferSize = 2 * 1024 * 1024;
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AVCodecEncoder::AVCodecEncoder(uint32 codecID, int bitRateScale)
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:
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Encoder(),
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fBitRateScale(bitRateScale),
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fCodecID((enum CodecID)codecID),
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fCodec(NULL),
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fOwnContext(avcodec_alloc_context()),
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fContext(fOwnContext),
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fCodecInitStatus(CODEC_INIT_NEEDED),
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fFrame(avcodec_alloc_frame()),
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fSwsContext(NULL),
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fFramesWritten(0),
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fChunkBuffer(new(std::nothrow) uint8[kDefaultChunkBufferSize])
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{
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TRACE("AVCodecEncoder::AVCodecEncoder()\n");
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if (fCodecID > 0) {
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fCodec = avcodec_find_encoder(fCodecID);
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TRACE(" found AVCodec for %u: %p\n", fCodecID, fCodec);
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}
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memset(&fInputFormat, 0, sizeof(media_format));
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fAudioFifo = av_fifo_alloc(0);
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fDstFrame.data[0] = NULL;
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fDstFrame.data[1] = NULL;
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fDstFrame.data[2] = NULL;
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fDstFrame.data[3] = NULL;
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fDstFrame.linesize[0] = 0;
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fDstFrame.linesize[1] = 0;
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fDstFrame.linesize[2] = 0;
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fDstFrame.linesize[3] = 0;
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// Initial parameters, so we know if the user changed them
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fEncodeParameters.avg_field_size = 0;
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fEncodeParameters.max_field_size = 0;
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fEncodeParameters.quality = 1.0f;
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}
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AVCodecEncoder::~AVCodecEncoder()
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{
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TRACE("AVCodecEncoder::~AVCodecEncoder()\n");
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_CloseCodecIfNeeded();
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if (fSwsContext != NULL)
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sws_freeContext(fSwsContext);
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av_fifo_free(fAudioFifo);
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avpicture_free(&fDstFrame);
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// NOTE: Do not use avpicture_free() on fSrcFrame!! We fill the picture
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// data on the fly with the media buffer data passed to Encode().
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if (fFrame != NULL) {
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fFrame->data[0] = NULL;
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fFrame->data[1] = NULL;
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fFrame->data[2] = NULL;
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fFrame->data[3] = NULL;
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fFrame->linesize[0] = 0;
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fFrame->linesize[1] = 0;
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fFrame->linesize[2] = 0;
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fFrame->linesize[3] = 0;
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av_free(fFrame);
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}
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av_free(fOwnContext);
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delete[] fChunkBuffer;
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}
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status_t
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AVCodecEncoder::AcceptedFormat(const media_format* proposedInputFormat,
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media_format* _acceptedInputFormat)
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{
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TRACE("AVCodecEncoder::AcceptedFormat(%p, %p)\n", proposedInputFormat,
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_acceptedInputFormat);
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if (proposedInputFormat == NULL)
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return B_BAD_VALUE;
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if (_acceptedInputFormat != NULL) {
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memcpy(_acceptedInputFormat, proposedInputFormat,
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sizeof(media_format));
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}
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return B_OK;
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}
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status_t
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AVCodecEncoder::SetUp(const media_format* inputFormat)
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{
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TRACE("AVCodecEncoder::SetUp()\n");
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if (fContext == NULL)
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return B_NO_INIT;
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if (inputFormat == NULL)
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return B_BAD_VALUE;
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// Codec IDs for raw-formats may need to be figured out here.
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if (fCodec == NULL && fCodecID == CODEC_ID_NONE) {
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fCodecID = raw_audio_codec_id_for(*inputFormat);
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if (fCodecID != CODEC_ID_NONE)
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fCodec = avcodec_find_encoder(fCodecID);
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}
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if (fCodec == NULL) {
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TRACE(" encoder not found!\n");
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return B_NO_INIT;
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}
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_CloseCodecIfNeeded();
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fInputFormat = *inputFormat;
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fFramesWritten = 0;
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const uchar* userData = inputFormat->user_data;
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if (*(uint32*)userData == 'ffmp') {
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userData += sizeof(uint32);
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// The Writer plugin used is the FFmpeg plugin. It stores the
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// AVCodecContext pointer in the user data section. Use this
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// context instead of our own. It requires the Writer living in
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// the same team, of course.
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app_info appInfo;
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if (be_app->GetAppInfo(&appInfo) == B_OK
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&& *(team_id*)userData == appInfo.team) {
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userData += sizeof(team_id);
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// Use the AVCodecContext from the Writer. This works better
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// than using our own context with some encoders.
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fContext = *(AVCodecContext**)userData;
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}
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}
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return _Setup();
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}
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status_t
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AVCodecEncoder::GetEncodeParameters(encode_parameters* parameters) const
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{
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TRACE("AVCodecEncoder::GetEncodeParameters(%p)\n", parameters);
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// TODO: Implement maintaining an automatically calculated bit_rate versus
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// a user specified (via SetEncodeParameters()) bit_rate. At this point, the
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// fContext->bit_rate may not yet have been specified (_Setup() was never
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// called yet). So it cannot work like the code below, but in any case, it's
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// showing how to convert between the values (albeit untested).
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// int avgBytesPerSecond = fContext->bit_rate / 8;
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// int maxBytesPerSecond = (fContext->bit_rate
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// + fContext->bit_rate_tolerance) / 8;
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//
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// if (fInputFormat.type == B_MEDIA_RAW_AUDIO) {
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// fEncodeParameters.avg_field_size = (int32)(avgBytesPerSecond
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// / fInputFormat.u.raw_audio.frame_rate);
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// fEncodeParameters.max_field_size = (int32)(maxBytesPerSecond
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// / fInputFormat.u.raw_audio.frame_rate);
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// } else if (fInputFormat.type == B_MEDIA_RAW_VIDEO) {
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// fEncodeParameters.avg_field_size = (int32)(avgBytesPerSecond
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// / fInputFormat.u.raw_video.field_rate);
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// fEncodeParameters.max_field_size = (int32)(maxBytesPerSecond
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// / fInputFormat.u.raw_video.field_rate);
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// }
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parameters->quality = fEncodeParameters.quality;
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return B_OK;
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}
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status_t
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AVCodecEncoder::SetEncodeParameters(encode_parameters* parameters)
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{
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TRACE("AVCodecEncoder::SetEncodeParameters(%p)\n", parameters);
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if (fFramesWritten > 0)
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return B_NOT_SUPPORTED;
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fEncodeParameters.quality = parameters->quality;
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TRACE(" quality: %.5f\n", parameters->quality);
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if (fEncodeParameters.quality == 0.0f) {
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TRACE(" using default quality (1.0)\n");
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fEncodeParameters.quality = 1.0f;
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}
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// TODO: Auto-bit_rate versus user supplied. See above.
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// int avgBytesPerSecond = 0;
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// int maxBytesPerSecond = 0;
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//
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// if (fInputFormat.type == B_MEDIA_RAW_AUDIO) {
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// avgBytesPerSecond = (int)(parameters->avg_field_size
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// * fInputFormat.u.raw_audio.frame_rate);
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// maxBytesPerSecond = (int)(parameters->max_field_size
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// * fInputFormat.u.raw_audio.frame_rate);
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// } else if (fInputFormat.type == B_MEDIA_RAW_VIDEO) {
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// avgBytesPerSecond = (int)(parameters->avg_field_size
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// * fInputFormat.u.raw_video.field_rate);
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// maxBytesPerSecond = (int)(parameters->max_field_size
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// * fInputFormat.u.raw_video.field_rate);
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// }
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//
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// if (maxBytesPerSecond < avgBytesPerSecond)
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// maxBytesPerSecond = avgBytesPerSecond;
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//
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// // Reset these, so we can tell the difference between uninitialized
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// // and initialized...
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// if (avgBytesPerSecond > 0) {
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// fContext->bit_rate = avgBytesPerSecond * 8;
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// fContext->bit_rate_tolerance = (maxBytesPerSecond
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// - avgBytesPerSecond) * 8;
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// fBitRateControlledByUser = true;
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// }
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return _Setup();
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}
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status_t
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AVCodecEncoder::Encode(const void* buffer, int64 frameCount,
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media_encode_info* info)
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{
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TRACE("AVCodecEncoder::Encode(%p, %lld, %p)\n", buffer, frameCount, info);
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if (!_OpenCodecIfNeeded())
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return B_NO_INIT;
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if (fInputFormat.type == B_MEDIA_RAW_AUDIO)
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return _EncodeAudio(buffer, frameCount, info);
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else if (fInputFormat.type == B_MEDIA_RAW_VIDEO)
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return _EncodeVideo(buffer, frameCount, info);
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else
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return B_NO_INIT;
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}
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// #pragma mark -
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status_t
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AVCodecEncoder::_Setup()
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{
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TRACE("AVCodecEncoder::_Setup\n");
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int rawBitRate;
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if (fInputFormat.type == B_MEDIA_RAW_VIDEO) {
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TRACE(" B_MEDIA_RAW_VIDEO\n");
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// frame rate
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fContext->time_base.den = (int)fInputFormat.u.raw_video.field_rate;
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fContext->time_base.num = 1;
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// video size
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fContext->width = fInputFormat.u.raw_video.display.line_width;
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fContext->height = fInputFormat.u.raw_video.display.line_count;
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fContext->gop_size = 12;
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// TODO: Fix pixel format or setup conversion method...
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for (int i = 0; fCodec->pix_fmts[i] != PIX_FMT_NONE; i++) {
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// Use the last supported pixel format, which we hope is the
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// one with the best quality.
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fContext->pix_fmt = fCodec->pix_fmts[i];
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}
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// TODO: Setup rate control:
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// fContext->rate_emu = 0;
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// fContext->rc_eq = NULL;
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// fContext->rc_max_rate = 0;
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// fContext->rc_min_rate = 0;
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// TODO: Try to calculate a good bit rate...
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rawBitRate = (int)(fContext->width * fContext->height * 2
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* fInputFormat.u.raw_video.field_rate) * 8;
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// Pixel aspect ratio
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fContext->sample_aspect_ratio.num
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= fInputFormat.u.raw_video.pixel_width_aspect;
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fContext->sample_aspect_ratio.den
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= fInputFormat.u.raw_video.pixel_height_aspect;
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if (fContext->sample_aspect_ratio.num == 0
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|| fContext->sample_aspect_ratio.den == 0) {
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av_reduce(&fContext->sample_aspect_ratio.num,
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&fContext->sample_aspect_ratio.den, fContext->width,
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fContext->height, 255);
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}
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// TODO: This should already happen in AcceptFormat()
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if (fInputFormat.u.raw_video.display.bytes_per_row == 0) {
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fInputFormat.u.raw_video.display.bytes_per_row
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= fContext->width * 4;
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}
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fFrame->pts = 0;
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// Allocate space for colorspace converted AVPicture
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// TODO: Check allocations...
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avpicture_alloc(&fDstFrame, fContext->pix_fmt, fContext->width,
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fContext->height);
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// Make the frame point to the data in the converted AVPicture
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fFrame->data[0] = fDstFrame.data[0];
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fFrame->data[1] = fDstFrame.data[1];
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fFrame->data[2] = fDstFrame.data[2];
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fFrame->data[3] = fDstFrame.data[3];
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fFrame->linesize[0] = fDstFrame.linesize[0];
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fFrame->linesize[1] = fDstFrame.linesize[1];
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fFrame->linesize[2] = fDstFrame.linesize[2];
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fFrame->linesize[3] = fDstFrame.linesize[3];
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fSwsContext = sws_getContext(fContext->width, fContext->height,
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colorspace_to_pixfmt(fInputFormat.u.raw_video.display.format),
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fContext->width, fContext->height,
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fContext->pix_fmt, SWS_FAST_BILINEAR, NULL, NULL, NULL);
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} else if (fInputFormat.type == B_MEDIA_RAW_AUDIO) {
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TRACE(" B_MEDIA_RAW_AUDIO\n");
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// frame rate
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fContext->sample_rate = (int)fInputFormat.u.raw_audio.frame_rate;
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// channels
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fContext->channels = fInputFormat.u.raw_audio.channel_count;
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// raw bitrate
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rawBitRate = fContext->sample_rate * fContext->channels
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* (fInputFormat.u.raw_audio.format
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& media_raw_audio_format::B_AUDIO_SIZE_MASK) * 8;
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// sample format
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switch (fInputFormat.u.raw_audio.format) {
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case media_raw_audio_format::B_AUDIO_FLOAT:
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fContext->sample_fmt = SAMPLE_FMT_FLT;
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break;
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case media_raw_audio_format::B_AUDIO_DOUBLE:
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fContext->sample_fmt = SAMPLE_FMT_DBL;
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break;
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case media_raw_audio_format::B_AUDIO_INT:
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fContext->sample_fmt = SAMPLE_FMT_S32;
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break;
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case media_raw_audio_format::B_AUDIO_SHORT:
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fContext->sample_fmt = SAMPLE_FMT_S16;
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break;
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case media_raw_audio_format::B_AUDIO_UCHAR:
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fContext->sample_fmt = SAMPLE_FMT_U8;
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break;
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case media_raw_audio_format::B_AUDIO_CHAR:
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default:
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return B_MEDIA_BAD_FORMAT;
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break;
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}
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if (fInputFormat.u.raw_audio.channel_mask == 0) {
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// guess the channel mask...
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switch (fInputFormat.u.raw_audio.channel_count) {
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default:
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case 2:
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fContext->channel_layout = CH_LAYOUT_STEREO;
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break;
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case 1:
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fContext->channel_layout = CH_LAYOUT_MONO;
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break;
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case 3:
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fContext->channel_layout = CH_LAYOUT_SURROUND;
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break;
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case 4:
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fContext->channel_layout = CH_LAYOUT_QUAD;
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break;
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case 5:
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fContext->channel_layout = CH_LAYOUT_5POINT0;
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break;
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case 6:
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fContext->channel_layout = CH_LAYOUT_5POINT1;
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break;
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case 8:
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fContext->channel_layout = CH_LAYOUT_7POINT1;
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break;
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case 10:
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fContext->channel_layout = CH_LAYOUT_7POINT1_WIDE;
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break;
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}
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} else {
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// The bits match 1:1 for media_multi_channels and FFmpeg defines.
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fContext->channel_layout = fInputFormat.u.raw_audio.channel_mask;
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}
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} else {
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TRACE(" UNSUPPORTED MEDIA TYPE!\n");
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return B_NOT_SUPPORTED;
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}
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// TODO: Support letting the user overwrite this via
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// SetEncodeParameters(). See comments there...
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int wantedBitRate = (int)(rawBitRate / fBitRateScale
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* fEncodeParameters.quality);
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if (wantedBitRate == 0)
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wantedBitRate = (int)(rawBitRate / fBitRateScale);
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fContext->bit_rate = wantedBitRate;
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if (fInputFormat.type == B_MEDIA_RAW_AUDIO) {
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// Some audio encoders support certain bitrates only. Use the
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// closest match to the wantedBitRate.
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const int kBitRates[] = {
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32000, 40000, 48000, 56000, 64000, 80000, 96000, 112000, 128000,
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160000, 192000, 224000, 256000, 320000, 384000, 448000, 512000,
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576000, 640000
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};
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int diff = wantedBitRate;
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for (int i = 0; i < sizeof(kBitRates) / sizeof(int); i++) {
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int currentDiff = abs(wantedBitRate - kBitRates[i]);
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if (currentDiff < diff) {
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fContext->bit_rate = kBitRates[i];
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diff = currentDiff;
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} else
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break;
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}
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}
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TRACE(" rawBitRate: %d, wantedBitRate: %d (%.1f), "
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"context bitrate: %d\n", rawBitRate, wantedBitRate,
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fEncodeParameters.quality, fContext->bit_rate);
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// Add some known fixes from the FFmpeg API example:
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if (fContext->codec_id == CODEC_ID_MPEG2VIDEO) {
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// Just for testing, we also add B frames */
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fContext->max_b_frames = 2;
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} else if (fContext->codec_id == CODEC_ID_MPEG1VIDEO){
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// Needed to avoid using macroblocks in which some coeffs overflow.
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// This does not happen with normal video, it just happens here as
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// the motion of the chroma plane does not match the luma plane.
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fContext->mb_decision = 2;
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}
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// Unfortunately, we may fail later, when we try to open the codec
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// for real... but we need to delay this because we still allow
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// parameter/quality changes.
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return B_OK;
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}
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bool
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AVCodecEncoder::_OpenCodecIfNeeded()
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{
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if (fContext != fOwnContext) {
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// We are using the AVCodecContext of the AVFormatWriter plugin,
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// and don't maintain it's open/close state.
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return true;
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}
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if (fCodecInitStatus == CODEC_INIT_DONE)
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return true;
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if (fCodecInitStatus == CODEC_INIT_FAILED)
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return false;
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// Open the codec
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int result = avcodec_open(fContext, fCodec);
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if (result >= 0)
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|
fCodecInitStatus = CODEC_INIT_DONE;
|
|
else
|
|
fCodecInitStatus = CODEC_INIT_FAILED;
|
|
|
|
TRACE(" avcodec_open(%p, %p): %d\n", fContext, fCodec, result);
|
|
|
|
return fCodecInitStatus == CODEC_INIT_DONE;
|
|
|
|
}
|
|
|
|
|
|
void
|
|
AVCodecEncoder::_CloseCodecIfNeeded()
|
|
{
|
|
if (fContext != fOwnContext) {
|
|
// See _OpenCodecIfNeeded().
|
|
return;
|
|
}
|
|
|
|
if (fCodecInitStatus == CODEC_INIT_DONE) {
|
|
avcodec_close(fContext);
|
|
fCodecInitStatus = CODEC_INIT_NEEDED;
|
|
}
|
|
}
|
|
|
|
|
|
static const int64 kNoPTSValue = 0x8000000000000000LL;
|
|
// NOTE: For some reasons, I have trouble with the avcodec.h define:
|
|
// #define AV_NOPTS_VALUE INT64_C(0x8000000000000000)
|
|
// INT64_C is not defined here.
|
|
|
|
status_t
|
|
AVCodecEncoder::_EncodeAudio(const void* _buffer, int64 frameCount,
|
|
media_encode_info* info)
|
|
{
|
|
TRACE("AVCodecEncoder::_EncodeAudio(%p, %lld, %p)\n", _buffer, frameCount,
|
|
info);
|
|
|
|
if (fChunkBuffer == NULL)
|
|
return B_NO_MEMORY;
|
|
|
|
status_t ret = B_OK;
|
|
|
|
const uint8* buffer = reinterpret_cast<const uint8*>(_buffer);
|
|
|
|
size_t inputSampleSize = fInputFormat.u.raw_audio.format
|
|
& media_raw_audio_format::B_AUDIO_SIZE_MASK;
|
|
size_t inputFrameSize = inputSampleSize
|
|
* fInputFormat.u.raw_audio.channel_count;
|
|
|
|
size_t bufferSize = frameCount * inputFrameSize;
|
|
bufferSize = min_c(bufferSize, kDefaultChunkBufferSize);
|
|
|
|
if (fContext->frame_size > 1) {
|
|
// Encoded audio. Things work differently from raw audio. We need
|
|
// the fAudioFifo to pipe data.
|
|
if (av_fifo_realloc2(fAudioFifo,
|
|
av_fifo_size(fAudioFifo) + bufferSize) < 0) {
|
|
TRACE(" av_fifo_realloc2() failed\n");
|
|
return B_NO_MEMORY;
|
|
}
|
|
av_fifo_generic_write(fAudioFifo, const_cast<uint8*>(buffer),
|
|
bufferSize, NULL);
|
|
|
|
int frameBytes = fContext->frame_size * inputFrameSize;
|
|
uint8* tempBuffer = new(std::nothrow) uint8[frameBytes];
|
|
if (tempBuffer == NULL)
|
|
return B_NO_MEMORY;
|
|
|
|
// Encode as many chunks as can be read from the FIFO.
|
|
while (av_fifo_size(fAudioFifo) >= frameBytes) {
|
|
av_fifo_generic_read(fAudioFifo, tempBuffer, frameBytes, NULL);
|
|
|
|
ret = _EncodeAudio(tempBuffer, frameBytes, fContext->frame_size,
|
|
info);
|
|
if (ret != B_OK)
|
|
break;
|
|
}
|
|
|
|
delete[] tempBuffer;
|
|
} else {
|
|
// Raw audio. The number of bytes returned from avcodec_encode_audio()
|
|
// is always the same as the number of input bytes.
|
|
return _EncodeAudio(buffer, bufferSize, frameCount,
|
|
info);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
|
|
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));
|
|
|
|
if (usedBytes < 0) {
|
|
TRACE(" avcodec_encode_audio() failed: %d\n", usedBytes);
|
|
return B_ERROR;
|
|
}
|
|
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);
|
|
// }
|
|
|
|
// 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;
|
|
}
|
|
|
|
fFramesWritten += frameCount;
|
|
|
|
return B_OK;
|
|
}
|
|
|
|
|
|
status_t
|
|
AVCodecEncoder::_EncodeVideo(const void* buffer, int64 frameCount,
|
|
media_encode_info* info)
|
|
{
|
|
TRACE_IO("AVCodecEncoder::_EncodeVideo(%p, %lld, %p)\n", buffer, frameCount,
|
|
info);
|
|
|
|
if (fChunkBuffer == NULL)
|
|
return B_NO_MEMORY;
|
|
|
|
status_t ret = B_OK;
|
|
|
|
while (frameCount > 0) {
|
|
size_t bpr = fInputFormat.u.raw_video.display.bytes_per_row;
|
|
size_t bufferSize = fInputFormat.u.raw_video.display.line_count * bpr;
|
|
|
|
// We should always get chunky bitmaps, so this code should be safe.
|
|
fSrcFrame.data[0] = (uint8_t*)buffer;
|
|
fSrcFrame.linesize[0] = bpr;
|
|
|
|
// Run the pixel format conversion
|
|
sws_scale(fSwsContext, fSrcFrame.data, fSrcFrame.linesize, 0,
|
|
fInputFormat.u.raw_video.display.line_count, fDstFrame.data,
|
|
fDstFrame.linesize);
|
|
|
|
// Encode one video chunk/frame.
|
|
int usedBytes = avcodec_encode_video(fContext, fChunkBuffer,
|
|
kDefaultChunkBufferSize, fFrame);
|
|
|
|
// avcodec.h says we need to set it.
|
|
fFrame->pts++;
|
|
|
|
if (usedBytes < 0) {
|
|
TRACE(" avcodec_encode_video() failed: %d\n", usedBytes);
|
|
return B_ERROR;
|
|
}
|
|
|
|
// 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);
|
|
}
|
|
|
|
// Setup media_encode_info, most important is the time stamp.
|
|
info->start_time = (bigtime_t)(fFramesWritten * 1000000LL
|
|
/ fInputFormat.u.raw_video.field_rate);
|
|
|
|
info->flags = 0;
|
|
if (fContext->coded_frame->key_frame)
|
|
info->flags |= B_MEDIA_KEY_FRAME;
|
|
|
|
// Write the chunk
|
|
ret = WriteChunk(fChunkBuffer, usedBytes, info);
|
|
if (ret != B_OK) {
|
|
TRACE(" error writing chunk: %s\n", strerror(ret));
|
|
break;
|
|
}
|
|
|
|
// Skip to the next frame (but usually, there is only one to encode
|
|
// for video).
|
|
frameCount--;
|
|
fFramesWritten++;
|
|
buffer = (const void*)((const uint8*)buffer + bufferSize);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|