blender/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp

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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 = 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(codec->sample_fmts) {
// Check if the prefered sample format for this codec is supported.
const enum AVSampleFormat *p = codec->sample_fmts;
for(; *p != -1; p++) {
if(*p == m_stream->codec->sample_fmt)
break;
}
if(*p == -1) {
// Sample format incompatible with codec. Defaulting to a format known to work.
m_stream->codec->sample_fmt = codec->sample_fmts[0];
}
}
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_sample_size = av_get_bytes_per_sample(m_codecCtx->sample_fmt);
m_frame_pts = 0;
m_deinterleave = av_sample_fmt_is_planar(m_codecCtx->sample_fmt);
if(m_deinterleave)
m_deinterleave_buffer.resize(m_input_size * m_codecCtx->channels * m_sample_size);
#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
if(m_deinterleave) {
for(int channel = 0; channel < m_codecCtx->channels; channel++) {
for(int i = 0; i < m_frame->nb_samples; i++) {
memcpy(reinterpret_cast<uint8_t*>(m_deinterleave_buffer.getBuffer()) + (i + channel * m_frame->nb_samples) * m_sample_size,
reinterpret_cast<uint8_t*>(data) + (m_codecCtx->channels * i + channel) * m_sample_size, m_sample_size);
}
}
data = m_deinterleave_buffer.getBuffer();
}
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;
}
}