blender/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
Sergey Sharybin e82b5f61fe Attempt to fix mixdown when building without sndfile
This case was handled specially in writeffmpeg.c and seems it makes
audio export happy in all cases now.

TODO: libav-10 doesn't work with AC3 codec yet because this bloody
library ONLY supports FLTP format and FFmpeg ONLY supports FLT.

This is not fun guy, it really isn't! Where's your conscience??

CC: nexyon
2014-02-28 14:51:03 +06:00

367 lines
10 KiB
C++

/*
* ***** 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 = AV_CODEC_ID_AAC;
break;
case AUD_CODEC_AC3:
m_outputFmt->audio_codec = AV_CODEC_ID_AC3;
break;
case AUD_CODEC_FLAC:
m_outputFmt->audio_codec = AV_CODEC_ID_FLAC;
break;
case AUD_CODEC_MP2:
m_outputFmt->audio_codec = AV_CODEC_ID_MP2;
break;
case AUD_CODEC_MP3:
m_outputFmt->audio_codec = AV_CODEC_ID_MP3;
break;
case AUD_CODEC_PCM:
switch(specs.format)
{
case AUD_FORMAT_U8:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_U8;
break;
case AUD_FORMAT_S16:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S16LE;
break;
case AUD_FORMAT_S24:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S24LE;
break;
case AUD_FORMAT_S32:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S32LE;
break;
case AUD_FORMAT_FLOAT32:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F32LE;
break;
case AUD_FORMAT_FLOAT64:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F64LE;
break;
default:
m_outputFmt->audio_codec = AV_CODEC_ID_NONE;
break;
}
break;
case AUD_CODEC_VORBIS:
m_outputFmt->audio_codec = AV_CODEC_ID_VORBIS;
break;
default:
m_outputFmt->audio_codec = AV_CODEC_ID_NONE;
break;
}
try
{
if(m_outputFmt->audio_codec == AV_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 = FF_MIN_BUFFER_SIZE * 8 / m_codecCtx->bits_per_coded_sample / m_codecCtx->channels;
m_input_buffer.resize(m_input_size * samplesize);
}
else
{
m_input_buffer.resize(m_codecCtx->frame_size * samplesize);
m_input_size = m_codecCtx->frame_size;
}
#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
m_frame = av_frame_alloc();
if (!m_frame)
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
avcodec_get_frame_defaults(m_frame);
m_frame->linesize[0] = m_input_size * samplesize;
m_frame->format = m_codecCtx->sample_fmt;
m_frame->nb_samples = m_input_size;
# ifdef FFMPEG_HAVE_AVFRAME_SAMPLE_RATE
m_frame->sample_rate = m_codecCtx->sample_rate;
# endif
# ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
m_frame->channel_layout = m_codecCtx->channel_layout;
# endif
m_audio_sample_size = av_get_bytes_per_sample(m_codecCtx->sample_fmt);
m_frame_pts = 0;
#endif
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]);
#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
av_frame_free(&m_frame);
#endif
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)
{
// 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 = { 0 };
av_init_packet(&packet);
#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
int got_output, ret;
m_frame->pts = m_frame_pts / av_q2d(m_codecCtx->time_base);
m_frame_pts++;
#ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
m_frame->channel_layout = m_codecCtx->channel_layout;
#endif
avcodec_fill_audio_frame(m_frame, m_codecCtx->channels, m_codecCtx->sample_fmt, reinterpret_cast<uint8_t*>(data),
m_frame->nb_samples * av_get_bytes_per_sample(m_codecCtx->sample_fmt) * m_codecCtx->channels, 1);
ret = avcodec_encode_audio2(m_codecCtx, &packet, m_frame, &got_output);
if(ret < 0)
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
if(!got_output)
return;
#else
sample_t* outbuf = m_output_buffer.getBuffer();
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.data = reinterpret_cast<uint8_t*>(outbuf);
#endif
if(packet.pts != AV_NOPTS_VALUE)
packet.pts = av_rescale_q(packet.pts, m_codecCtx->time_base, m_stream->time_base);
if(packet.dts != AV_NOPTS_VALUE)
packet.dts = av_rescale_q(packet.dts, m_codecCtx->time_base, m_stream->time_base);
if(packet.duration > 0)
packet.duration = av_rescale_q(packet.duration, m_codecCtx->time_base, m_stream->time_base);
packet.stream_index = m_stream->index;
packet.flags |= AV_PKT_FLAG_KEY;
if(av_interleaved_write_frame(m_formatCtx, &packet)) {
av_free_packet(&packet);
AUD_THROW(AUD_ERROR_FFMPEG, write_error);
}
av_free_packet(&packet);
}
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;
}
}