0b03785eb5
While it's not something we'll be using for the official release, it's nice to support new libraries at least on "it compiles" level, so it's not that many frustrated developers around. Nexyon, please have a look into Audaspace changes :)
396 lines
11 KiB
C++
396 lines
11 KiB
C++
/*
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* ***** BEGIN GPL LICENSE BLOCK *****
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*
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* Copyright 2009-2011 Jörg Hermann Müller
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*
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* This file is part of AudaSpace.
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*
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* Audaspace is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* AudaSpace is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with Audaspace; if not, write to the Free Software Foundation,
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* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
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*
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* ***** END GPL LICENSE BLOCK *****
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*/
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/** \file audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
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* \ingroup audffmpeg
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*/
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// needed for INT64_C
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#ifndef __STDC_CONSTANT_MACROS
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#define __STDC_CONSTANT_MACROS
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#endif
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#include "AUD_FFMPEGWriter.h"
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extern "C" {
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libavformat/avio.h>
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#include "ffmpeg_compat.h"
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}
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static const char* context_error = "AUD_FFMPEGWriter: Couldn't allocate context.";
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static const char* codec_error = "AUD_FFMPEGWriter: Invalid codec or codec not found.";
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static const char* stream_error = "AUD_FFMPEGWriter: Couldn't allocate stream.";
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static const char* format_error = "AUD_FFMPEGWriter: Unsupported sample format.";
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static const char* file_error = "AUD_FFMPEGWriter: File couldn't be written.";
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static const char* write_error = "AUD_FFMPEGWriter: Error writing packet.";
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AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename, AUD_DeviceSpecs specs, AUD_Container format, AUD_Codec codec, unsigned int bitrate) :
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m_position(0),
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m_specs(specs),
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m_input_samples(0)
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{
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static const char* formats[] = { NULL, "ac3", "flac", "matroska", "mp2", "mp3", "ogg", "wav" };
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m_formatCtx = avformat_alloc_context();
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if (!m_formatCtx) AUD_THROW(AUD_ERROR_FFMPEG, context_error);
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strcpy(m_formatCtx->filename, filename.c_str());
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m_outputFmt = m_formatCtx->oformat = av_guess_format(formats[format], filename.c_str(), NULL);
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if (!m_outputFmt) {
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avformat_free_context(m_formatCtx);
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AUD_THROW(AUD_ERROR_FFMPEG, context_error);
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}
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switch(codec)
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{
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case AUD_CODEC_AAC:
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m_outputFmt->audio_codec = AV_CODEC_ID_AAC;
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break;
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case AUD_CODEC_AC3:
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m_outputFmt->audio_codec = AV_CODEC_ID_AC3;
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break;
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case AUD_CODEC_FLAC:
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m_outputFmt->audio_codec = AV_CODEC_ID_FLAC;
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break;
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case AUD_CODEC_MP2:
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m_outputFmt->audio_codec = AV_CODEC_ID_MP2;
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break;
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case AUD_CODEC_MP3:
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m_outputFmt->audio_codec = AV_CODEC_ID_MP3;
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break;
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case AUD_CODEC_PCM:
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switch(specs.format)
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{
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case AUD_FORMAT_U8:
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m_outputFmt->audio_codec = AV_CODEC_ID_PCM_U8;
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break;
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case AUD_FORMAT_S16:
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m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S16LE;
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break;
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case AUD_FORMAT_S24:
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m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S24LE;
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break;
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case AUD_FORMAT_S32:
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m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S32LE;
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break;
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case AUD_FORMAT_FLOAT32:
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m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F32LE;
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break;
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case AUD_FORMAT_FLOAT64:
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m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F64LE;
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break;
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default:
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m_outputFmt->audio_codec = AV_CODEC_ID_NONE;
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break;
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}
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break;
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case AUD_CODEC_VORBIS:
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m_outputFmt->audio_codec = AV_CODEC_ID_VORBIS;
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break;
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default:
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m_outputFmt->audio_codec = AV_CODEC_ID_NONE;
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break;
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}
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try
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{
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if(m_outputFmt->audio_codec == AV_CODEC_ID_NONE)
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AUD_THROW(AUD_ERROR_SPECS, codec_error);
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m_stream = avformat_new_stream(m_formatCtx, NULL);
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if(!m_stream)
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AUD_THROW(AUD_ERROR_FFMPEG, stream_error);
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m_codecCtx = m_stream->codec;
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m_codecCtx->codec_id = m_outputFmt->audio_codec;
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m_codecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
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m_codecCtx->bit_rate = bitrate;
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m_codecCtx->sample_rate = int(m_specs.rate);
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m_codecCtx->channels = m_specs.channels;
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m_codecCtx->time_base.num = 1;
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m_codecCtx->time_base.den = m_codecCtx->sample_rate;
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switch(m_specs.format)
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{
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case AUD_FORMAT_U8:
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m_convert = AUD_convert_float_u8;
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m_codecCtx->sample_fmt = AV_SAMPLE_FMT_U8;
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break;
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case AUD_FORMAT_S16:
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m_convert = AUD_convert_float_s16;
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m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
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break;
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case AUD_FORMAT_S32:
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m_convert = AUD_convert_float_s32;
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m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S32;
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break;
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case AUD_FORMAT_FLOAT32:
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m_convert = AUD_convert_copy<float>;
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m_codecCtx->sample_fmt = AV_SAMPLE_FMT_FLT;
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break;
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case AUD_FORMAT_FLOAT64:
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m_convert = AUD_convert_float_double;
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m_codecCtx->sample_fmt = AV_SAMPLE_FMT_DBL;
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break;
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default:
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AUD_THROW(AUD_ERROR_FFMPEG, format_error);
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}
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try
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{
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if(m_formatCtx->oformat->flags & AVFMT_GLOBALHEADER)
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m_codecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
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AVCodec* codec = avcodec_find_encoder(m_codecCtx->codec_id);
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if(!codec)
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AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
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if(codec->sample_fmts) {
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// Check if the preferred sample format for this codec is supported.
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const enum AVSampleFormat *p = codec->sample_fmts;
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for(; *p != -1; p++) {
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if(*p == m_stream->codec->sample_fmt)
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break;
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}
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if(*p == -1) {
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// Sample format incompatible with codec. Defaulting to a format known to work.
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m_stream->codec->sample_fmt = codec->sample_fmts[0];
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}
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}
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if(avcodec_open2(m_codecCtx, codec, NULL))
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AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
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m_output_buffer.resize(FF_MIN_BUFFER_SIZE);
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int samplesize = AUD_MAX(AUD_SAMPLE_SIZE(m_specs), AUD_DEVICE_SAMPLE_SIZE(m_specs));
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if(m_codecCtx->frame_size <= 1) {
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m_input_size = FF_MIN_BUFFER_SIZE * 8 / m_codecCtx->bits_per_coded_sample / m_codecCtx->channels;
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m_input_buffer.resize(m_input_size * samplesize);
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}
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else
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{
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m_input_buffer.resize(m_codecCtx->frame_size * samplesize);
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m_input_size = m_codecCtx->frame_size;
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}
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#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
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m_frame = av_frame_alloc();
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if (!m_frame)
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AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
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av_frame_unref(m_frame);
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m_frame->linesize[0] = m_input_size * samplesize;
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m_frame->format = m_codecCtx->sample_fmt;
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m_frame->nb_samples = m_input_size;
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# ifdef FFMPEG_HAVE_AVFRAME_SAMPLE_RATE
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m_frame->sample_rate = m_codecCtx->sample_rate;
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# endif
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# ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
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m_frame->channel_layout = m_codecCtx->channel_layout;
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# endif
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m_sample_size = av_get_bytes_per_sample(m_codecCtx->sample_fmt);
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m_frame_pts = 0;
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m_deinterleave = av_sample_fmt_is_planar(m_codecCtx->sample_fmt);
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if(m_deinterleave)
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m_deinterleave_buffer.resize(m_input_size * m_codecCtx->channels * m_sample_size);
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#endif
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try
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{
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if(avio_open(&m_formatCtx->pb, filename.c_str(), AVIO_FLAG_WRITE))
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AUD_THROW(AUD_ERROR_FILE, file_error);
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if(avformat_write_header(m_formatCtx, NULL) < 0) {
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throw;
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}
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}
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catch(AUD_Exception&)
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{
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avcodec_close(m_codecCtx);
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av_freep(&m_formatCtx->streams[0]->codec);
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throw;
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}
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}
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catch(AUD_Exception&)
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{
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av_freep(&m_formatCtx->streams[0]);
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throw;
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}
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}
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catch(AUD_Exception&)
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{
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av_free(m_formatCtx);
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throw;
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}
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}
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AUD_FFMPEGWriter::~AUD_FFMPEGWriter()
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{
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// writte missing data
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if(m_input_samples)
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{
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sample_t* buf = m_input_buffer.getBuffer();
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memset(buf + m_specs.channels * m_input_samples, 0,
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(m_input_size - m_input_samples) * AUD_DEVICE_SAMPLE_SIZE(m_specs));
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encode(buf);
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}
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av_write_trailer(m_formatCtx);
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avcodec_close(m_codecCtx);
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av_freep(&m_formatCtx->streams[0]->codec);
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av_freep(&m_formatCtx->streams[0]);
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#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
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av_frame_free(&m_frame);
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#endif
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avio_close(m_formatCtx->pb);
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av_free(m_formatCtx);
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}
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int AUD_FFMPEGWriter::getPosition() const
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{
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return m_position;
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}
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AUD_DeviceSpecs AUD_FFMPEGWriter::getSpecs() const
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{
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return m_specs;
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}
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void AUD_FFMPEGWriter::encode(sample_t* data)
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{
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// convert first
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if(m_input_size)
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m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(data), m_input_size * m_specs.channels);
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AVPacket packet = { 0 };
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av_init_packet(&packet);
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#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
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int got_output, ret;
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m_frame->pts = m_frame_pts / av_q2d(m_codecCtx->time_base);
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m_frame_pts++;
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#ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
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m_frame->channel_layout = m_codecCtx->channel_layout;
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#endif
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if(m_deinterleave) {
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for(int channel = 0; channel < m_codecCtx->channels; channel++) {
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for(int i = 0; i < m_frame->nb_samples; i++) {
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memcpy(reinterpret_cast<uint8_t*>(m_deinterleave_buffer.getBuffer()) + (i + channel * m_frame->nb_samples) * m_sample_size,
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reinterpret_cast<uint8_t*>(data) + (m_codecCtx->channels * i + channel) * m_sample_size, m_sample_size);
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}
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}
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data = m_deinterleave_buffer.getBuffer();
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}
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avcodec_fill_audio_frame(m_frame, m_codecCtx->channels, m_codecCtx->sample_fmt, reinterpret_cast<uint8_t*>(data),
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m_frame->nb_samples * av_get_bytes_per_sample(m_codecCtx->sample_fmt) * m_codecCtx->channels, 1);
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ret = avcodec_encode_audio2(m_codecCtx, &packet, m_frame, &got_output);
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if(ret < 0)
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AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
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if(!got_output)
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return;
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#else
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sample_t* outbuf = m_output_buffer.getBuffer();
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packet.size = avcodec_encode_audio(m_codecCtx, reinterpret_cast<uint8_t*>(outbuf), m_output_buffer.getSize(), reinterpret_cast<short*>(data));
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if(m_codecCtx->coded_frame && m_codecCtx->coded_frame->pts != AV_NOPTS_VALUE)
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packet.pts = av_rescale_q(m_codecCtx->coded_frame->pts, m_codecCtx->time_base, m_stream->time_base);
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packet.flags |= AV_PKT_FLAG_KEY;
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packet.data = reinterpret_cast<uint8_t*>(outbuf);
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#endif
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if(packet.pts != AV_NOPTS_VALUE)
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packet.pts = av_rescale_q(packet.pts, m_codecCtx->time_base, m_stream->time_base);
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if(packet.dts != AV_NOPTS_VALUE)
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packet.dts = av_rescale_q(packet.dts, m_codecCtx->time_base, m_stream->time_base);
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if(packet.duration > 0)
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packet.duration = av_rescale_q(packet.duration, m_codecCtx->time_base, m_stream->time_base);
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packet.stream_index = m_stream->index;
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packet.flags |= AV_PKT_FLAG_KEY;
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if(av_interleaved_write_frame(m_formatCtx, &packet)) {
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av_free_packet(&packet);
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AUD_THROW(AUD_ERROR_FFMPEG, write_error);
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}
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av_free_packet(&packet);
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}
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void AUD_FFMPEGWriter::write(unsigned int length, sample_t* buffer)
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{
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unsigned int samplesize = AUD_SAMPLE_SIZE(m_specs);
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if(m_input_size)
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{
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sample_t* inbuf = m_input_buffer.getBuffer();
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while(length)
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{
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unsigned int len = AUD_MIN(m_input_size - m_input_samples, length);
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memcpy(inbuf + m_input_samples * m_specs.channels, buffer, len * samplesize);
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buffer += len * m_specs.channels;
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m_input_samples += len;
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m_position += len;
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length -= len;
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if(m_input_samples == m_input_size)
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{
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encode(inbuf);
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m_input_samples = 0;
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}
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}
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}
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else // PCM data, can write directly!
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{
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int samplesize = AUD_SAMPLE_SIZE(m_specs);
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if(m_output_buffer.getSize() != length * m_specs.channels * m_codecCtx->bits_per_coded_sample / 8)
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m_output_buffer.resize(length * m_specs.channels * m_codecCtx->bits_per_coded_sample / 8);
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m_input_buffer.assureSize(length * AUD_MAX(AUD_DEVICE_SAMPLE_SIZE(m_specs), samplesize));
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sample_t* buf = m_input_buffer.getBuffer();
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m_convert(reinterpret_cast<data_t*>(buf), reinterpret_cast<data_t*>(buffer), length * m_specs.channels);
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encode(buf);
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m_position += length;
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}
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}
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