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blender/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
Sergey Sharybin aab01c86cf Patch #33837: ffmpeg1.1 and libav9.1 compatibility update 
Patch makes it possible to compile blender with recent ffmpeg
and libav libraries, mainly by getting rid of deprecated API.

Original patch by Campbell Barton with own modifications to
support compilation with older ffmpeg versions.

This patch could break compatibility of FFV1 videos playing
back in older players, mainly because of alpha support changes.
Preserving compatibility with such players became a headache
and think it's high time to get rid of workarounds here.
2013-01-12 12:51:10 +00:00

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/*
* ***** BEGIN GPL LICENSE BLOCK *****
*
* Copyright 2009-2011 Jörg Hermann Müller
*
* This file is part of AudaSpace.
*
* Audaspace is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* AudaSpace is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with Audaspace; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*
* ***** END GPL LICENSE BLOCK *****
*/
/** \file audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
* \ingroup audffmpeg
*/
// needed for INT64_C
#ifndef __STDC_CONSTANT_MACROS
#define __STDC_CONSTANT_MACROS
#endif
#include "AUD_FFMPEGWriter.h"
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include "ffmpeg_compat.h"
}
static const char* context_error = "AUD_FFMPEGWriter: Couldn't allocate context.";
static const char* codec_error = "AUD_FFMPEGWriter: Invalid codec or codec not found.";
static const char* stream_error = "AUD_FFMPEGWriter: Couldn't allocate stream.";
static const char* format_error = "AUD_FFMPEGWriter: Unsupported sample format.";
static const char* file_error = "AUD_FFMPEGWriter: File couldn't be written.";
static const char* write_error = "AUD_FFMPEGWriter: Error writing packet.";
AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename, AUD_DeviceSpecs specs, AUD_Container format, AUD_Codec codec, unsigned int bitrate) :
m_position(0),
m_specs(specs),
m_input_samples(0)
{
static const char* formats[] = { NULL, "ac3", "flac", "matroska", "mp2", "mp3", "ogg", "wav" };
m_formatCtx = avformat_alloc_context();
if (!m_formatCtx) AUD_THROW(AUD_ERROR_FFMPEG, context_error);
strcpy(m_formatCtx->filename, filename.c_str());
m_outputFmt = m_formatCtx->oformat = av_guess_format(formats[format], filename.c_str(), NULL);
if (!m_outputFmt) {
avformat_free_context(m_formatCtx);
AUD_THROW(AUD_ERROR_FFMPEG, context_error);
}
switch(codec)
{
case AUD_CODEC_AAC:
m_outputFmt->audio_codec = CODEC_ID_AAC;
break;
case AUD_CODEC_AC3:
m_outputFmt->audio_codec = CODEC_ID_AC3;
break;
case AUD_CODEC_FLAC:
m_outputFmt->audio_codec = CODEC_ID_FLAC;
break;
case AUD_CODEC_MP2:
m_outputFmt->audio_codec = CODEC_ID_MP2;
break;
case AUD_CODEC_MP3:
m_outputFmt->audio_codec = CODEC_ID_MP3;
break;
case AUD_CODEC_PCM:
switch(specs.format)
{
case AUD_FORMAT_U8:
m_outputFmt->audio_codec = CODEC_ID_PCM_U8;
break;
case AUD_FORMAT_S16:
m_outputFmt->audio_codec = CODEC_ID_PCM_S16LE;
break;
case AUD_FORMAT_S24:
m_outputFmt->audio_codec = CODEC_ID_PCM_S24LE;
break;
case AUD_FORMAT_S32:
m_outputFmt->audio_codec = CODEC_ID_PCM_S32LE;
break;
case AUD_FORMAT_FLOAT32:
m_outputFmt->audio_codec = CODEC_ID_PCM_F32LE;
break;
case AUD_FORMAT_FLOAT64:
m_outputFmt->audio_codec = CODEC_ID_PCM_F64LE;
break;
default:
m_outputFmt->audio_codec = CODEC_ID_NONE;
break;
}
break;
case AUD_CODEC_VORBIS:
m_outputFmt->audio_codec = CODEC_ID_VORBIS;
break;
default:
m_outputFmt->audio_codec = CODEC_ID_NONE;
break;
}
try
{
if(m_outputFmt->audio_codec == CODEC_ID_NONE)
AUD_THROW(AUD_ERROR_SPECS, codec_error);
m_stream = avformat_new_stream(m_formatCtx, NULL);
if(!m_stream)
AUD_THROW(AUD_ERROR_FFMPEG, stream_error);
m_codecCtx = m_stream->codec;
m_codecCtx->codec_id = m_outputFmt->audio_codec;
m_codecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
m_codecCtx->bit_rate = bitrate;
m_codecCtx->sample_rate = int(m_specs.rate);
m_codecCtx->channels = m_specs.channels;
m_codecCtx->time_base.num = 1;
m_codecCtx->time_base.den = m_codecCtx->sample_rate;
switch(m_specs.format)
{
case AUD_FORMAT_U8:
m_convert = AUD_convert_float_u8;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_U8;
break;
case AUD_FORMAT_S16:
m_convert = AUD_convert_float_s16;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
break;
case AUD_FORMAT_S32:
m_convert = AUD_convert_float_s32;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S32;
break;
case AUD_FORMAT_FLOAT32:
m_convert = AUD_convert_copy<float>;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_FLT;
break;
case AUD_FORMAT_FLOAT64:
m_convert = AUD_convert_float_double;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_DBL;
break;
default:
AUD_THROW(AUD_ERROR_FFMPEG, format_error);
}
try
{
if(m_formatCtx->oformat->flags & AVFMT_GLOBALHEADER)
m_codecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
AVCodec* codec = avcodec_find_encoder(m_codecCtx->codec_id);
if(!codec)
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
if(avcodec_open2(m_codecCtx, codec, NULL))
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
m_output_buffer.resize(FF_MIN_BUFFER_SIZE);
int samplesize = AUD_MAX(AUD_SAMPLE_SIZE(m_specs), AUD_DEVICE_SAMPLE_SIZE(m_specs));
if(m_codecCtx->frame_size <= 1)
m_input_size = 0;
else
{
m_input_buffer.resize(m_codecCtx->frame_size * samplesize);
m_input_size = m_codecCtx->frame_size;
}
try
{
if(avio_open(&m_formatCtx->pb, filename.c_str(), AVIO_FLAG_WRITE))
AUD_THROW(AUD_ERROR_FILE, file_error);
avformat_write_header(m_formatCtx, NULL);
}
catch(AUD_Exception&)
{
avcodec_close(m_codecCtx);
av_freep(&m_formatCtx->streams[0]->codec);
throw;
}
}
catch(AUD_Exception&)
{
av_freep(&m_formatCtx->streams[0]);
throw;
}
}
catch(AUD_Exception&)
{
av_free(m_formatCtx);
throw;
}
}
AUD_FFMPEGWriter::~AUD_FFMPEGWriter()
{
// writte missing data
if(m_input_samples)
{
sample_t* buf = m_input_buffer.getBuffer();
memset(buf + m_specs.channels * m_input_samples, 0,
(m_input_size - m_input_samples) * AUD_DEVICE_SAMPLE_SIZE(m_specs));
encode(buf);
}
av_write_trailer(m_formatCtx);
avcodec_close(m_codecCtx);
av_freep(&m_formatCtx->streams[0]->codec);
av_freep(&m_formatCtx->streams[0]);
avio_close(m_formatCtx->pb);
av_free(m_formatCtx);
}
int AUD_FFMPEGWriter::getPosition() const
{
return m_position;
}
AUD_DeviceSpecs AUD_FFMPEGWriter::getSpecs() const
{
return m_specs;
}
void AUD_FFMPEGWriter::encode(sample_t* data)
{
sample_t* outbuf = m_output_buffer.getBuffer();
// convert first
if(m_input_size)
m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(data), m_input_size * m_specs.channels);
AVPacket packet;
av_init_packet(&packet);
packet.size = avcodec_encode_audio(m_codecCtx, reinterpret_cast<uint8_t*>(outbuf), m_output_buffer.getSize(), reinterpret_cast<short*>(data));
if(m_codecCtx->coded_frame && m_codecCtx->coded_frame->pts != AV_NOPTS_VALUE)
packet.pts = av_rescale_q(m_codecCtx->coded_frame->pts, m_codecCtx->time_base, m_stream->time_base);
packet.flags |= AV_PKT_FLAG_KEY;
packet.stream_index = m_stream->index;
packet.data = reinterpret_cast<uint8_t*>(outbuf);
if(av_interleaved_write_frame(m_formatCtx, &packet))
AUD_THROW(AUD_ERROR_FFMPEG, write_error);
}
void AUD_FFMPEGWriter::write(unsigned int length, sample_t* buffer)
{
unsigned int samplesize = AUD_SAMPLE_SIZE(m_specs);
if(m_input_size)
{
sample_t* inbuf = m_input_buffer.getBuffer();
while(length)
{
unsigned int len = AUD_MIN(m_input_size - m_input_samples, length);
memcpy(inbuf + m_input_samples * m_specs.channels, buffer, len * samplesize);
buffer += len * m_specs.channels;
m_input_samples += len;
m_position += len;
length -= len;
if(m_input_samples == m_input_size)
{
encode(inbuf);
m_input_samples = 0;
}
}
}
else // PCM data, can write directly!
{
int samplesize = AUD_SAMPLE_SIZE(m_specs);
if(m_output_buffer.getSize() != length * m_specs.channels * m_codecCtx->bits_per_coded_sample / 8)
m_output_buffer.resize(length * m_specs.channels * m_codecCtx->bits_per_coded_sample / 8);
m_input_buffer.assureSize(length * AUD_MAX(AUD_DEVICE_SAMPLE_SIZE(m_specs), samplesize));
sample_t* buf = m_input_buffer.getBuffer();
m_convert(reinterpret_cast<data_t*>(buf), reinterpret_cast<data_t*>(buffer), length * m_specs.channels);
encode(buf);
m_position += length;
}
}
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