Changeset View
Changeset View
Standalone View
Standalone View
extern/audaspace/plugins/ffmpeg/FFMPEGWriter.cpp
- This file was added.
| /******************************************************************************* | |||||
| * Copyright 2009-2015 Jörg Müller | |||||
| * | |||||
| * Licensed under the Apache License, Version 2.0 (the "License"); | |||||
| * you may not use this file except in compliance with the License. | |||||
| * You may obtain a copy of the License at | |||||
| * | |||||
| * http://www.apache.org/licenses/LICENSE-2.0 | |||||
| * | |||||
| * Unless required by applicable law or agreed to in writing, software | |||||
| * distributed under the License is distributed on an "AS IS" BASIS, | |||||
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | |||||
| * See the License for the specific language governing permissions and | |||||
| * limitations under the License. | |||||
| ******************************************************************************/ | |||||
| #include "FFMPEGWriter.h" | |||||
| #include "Exception.h" | |||||
| #include <algorithm> | |||||
| #include <cstring> | |||||
| extern "C" { | |||||
| #include <libavcodec/avcodec.h> | |||||
| #include <libavformat/avio.h> | |||||
| } | |||||
| AUD_NAMESPACE_BEGIN | |||||
| void FFMPEGWriter::encode() | |||||
| { | |||||
| sample_t* data = m_input_buffer.getBuffer(); | |||||
| if(m_deinterleave) | |||||
| { | |||||
| m_deinterleave_buffer.assureSize(m_input_buffer.getSize()); | |||||
| sample_t* dbuf = m_deinterleave_buffer.getBuffer(); | |||||
| // deinterleave | |||||
| int single_size = AUD_FORMAT_SIZE(m_specs.format); | |||||
| for(int channel = 0; channel < m_specs.channels; channel++) | |||||
| { | |||||
| for(int i = 0; i < m_input_buffer.getSize() / AUD_SAMPLE_SIZE(m_specs); i++) | |||||
| { | |||||
| std::memcpy(((data_t*)dbuf) + (m_input_samples * channel + i) * single_size, | |||||
| ((data_t*)data) + ((m_specs.channels * i) + channel) * single_size, single_size); | |||||
| } | |||||
| } | |||||
| // convert first | |||||
| if(m_input_size) | |||||
| m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(dbuf), m_input_samples * m_specs.channels); | |||||
| else | |||||
| std::memcpy(data, dbuf, m_input_buffer.getSize()); | |||||
| } | |||||
| else | |||||
| // convert first | |||||
| if(m_input_size) | |||||
| m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(data), m_input_samples * m_specs.channels); | |||||
| AVPacket packet; | |||||
| packet.data = nullptr; | |||||
| packet.size = 0; | |||||
| av_init_packet(&packet); | |||||
| AVFrame* frame = av_frame_alloc(); | |||||
| av_frame_unref(frame); | |||||
| int got_packet; | |||||
| frame->nb_samples = m_input_samples; | |||||
| frame->format = m_codecCtx->sample_fmt; | |||||
| if(avcodec_fill_audio_frame(frame, m_specs.channels, m_codecCtx->sample_fmt, reinterpret_cast<data_t*>(data), m_input_samples * AUD_DEVICE_SAMPLE_SIZE(m_specs), 1) < 0) | |||||
| AUD_THROW(FileException, "File couldn't be written, filling the audio frame failed with ffmpeg."); | |||||
| if(m_codecCtx->coded_frame && m_codecCtx->coded_frame->pts != AV_NOPTS_VALUE) | |||||
| frame->pts = av_rescale_q(m_codecCtx->coded_frame->pts, m_codecCtx->time_base, m_stream->time_base); | |||||
| else | |||||
| frame->pts = AV_NOPTS_VALUE; | |||||
| if(avcodec_encode_audio2(m_codecCtx, &packet, frame, &got_packet)) | |||||
| { | |||||
| av_frame_free(&frame); | |||||
| AUD_THROW(FileException, "File couldn't be written, audio encoding failed with ffmpeg."); | |||||
| } | |||||
| if(got_packet) | |||||
| { | |||||
| packet.flags |= AV_PKT_FLAG_KEY; | |||||
| packet.stream_index = m_stream->index; | |||||
| if(av_write_frame(m_formatCtx, &packet) < 0) | |||||
| { | |||||
| av_free_packet(&packet); | |||||
| av_frame_free(&frame); | |||||
| AUD_THROW(FileException, "Frame couldn't be writen to the file with ffmpeg."); | |||||
| } | |||||
| av_free_packet(&packet); | |||||
| } | |||||
| av_frame_free(&frame); | |||||
| } | |||||
| void FFMPEGWriter::close() | |||||
| { | |||||
| int got_packet = true; | |||||
| while(got_packet) | |||||
| { | |||||
| AVPacket packet; | |||||
| packet.data = nullptr; | |||||
| packet.size = 0; | |||||
| av_init_packet(&packet); | |||||
| if(avcodec_encode_audio2(m_codecCtx, &packet, nullptr, &got_packet)) | |||||
| AUD_THROW(FileException, "File end couldn't be written, audio encoding failed with ffmpeg."); | |||||
| if(got_packet) | |||||
| { | |||||
| packet.flags |= AV_PKT_FLAG_KEY; | |||||
| packet.stream_index = m_stream->index; | |||||
| if(av_write_frame(m_formatCtx, &packet)) | |||||
| { | |||||
| av_free_packet(&packet); | |||||
| AUD_THROW(FileException, "Final frames couldn't be writen to the file with ffmpeg."); | |||||
| } | |||||
| av_free_packet(&packet); | |||||
| } | |||||
| } | |||||
| } | |||||
| FFMPEGWriter::FFMPEGWriter(std::string filename, DeviceSpecs specs, Container format, Codec codec, unsigned int bitrate) : | |||||
| m_position(0), | |||||
| m_specs(specs), | |||||
| m_input_samples(0), | |||||
| m_deinterleave(false) | |||||
| { | |||||
| static const char* formats[] = { nullptr, "ac3", "flac", "matroska", "mp2", "mp3", "ogg", "wav" }; | |||||
| if(avformat_alloc_output_context2(&m_formatCtx, nullptr, formats[format], filename.c_str()) < 0) | |||||
| AUD_THROW(FileException, "File couldn't be written, format couldn't be found with ffmpeg."); | |||||
| m_outputFmt = m_formatCtx->oformat; | |||||
| if(!m_outputFmt) { | |||||
| avformat_free_context(m_formatCtx); | |||||
| AUD_THROW(FileException, "File couldn't be written, output format couldn't be found with ffmpeg."); | |||||
| } | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_NONE; | |||||
| switch(codec) | |||||
| { | |||||
| case CODEC_AAC: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_AAC; | |||||
| break; | |||||
| case CODEC_AC3: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_AC3; | |||||
| break; | |||||
| case CODEC_FLAC: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_FLAC; | |||||
| break; | |||||
| case CODEC_MP2: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_MP2; | |||||
| break; | |||||
| case CODEC_MP3: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_MP3; | |||||
| break; | |||||
| case CODEC_OPUS: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_OPUS; | |||||
| break; | |||||
| case CODEC_PCM: | |||||
| switch(specs.format) | |||||
| { | |||||
| case FORMAT_U8: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_PCM_U8; | |||||
| break; | |||||
| case FORMAT_S16: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S16LE; | |||||
| break; | |||||
| case FORMAT_S24: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S24LE; | |||||
| break; | |||||
| case FORMAT_S32: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S32LE; | |||||
| break; | |||||
| case FORMAT_FLOAT32: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F32LE; | |||||
| break; | |||||
| case FORMAT_FLOAT64: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F64LE; | |||||
| break; | |||||
| default: | |||||
| m_outputFmt->audio_codec = AV_CODEC_ID_NONE; | |||||
| break; | |||||
| } | |||||
| break; | |||||
| case 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(FileException, "File couldn't be written, audio codec not found with ffmpeg."); | |||||
| AVCodec* codec = avcodec_find_encoder(m_outputFmt->audio_codec); | |||||
| if(!codec) | |||||
| AUD_THROW(FileException, "File couldn't be written, audio encoder couldn't be found with ffmpeg."); | |||||
| m_stream = avformat_new_stream(m_formatCtx, codec); | |||||
| if(!m_stream) | |||||
| AUD_THROW(FileException, "File couldn't be written, stream creation failed with ffmpeg."); | |||||
| m_stream->id = m_formatCtx->nb_streams - 1; | |||||
| m_codecCtx = m_stream->codec; | |||||
| switch(m_specs.format) | |||||
| { | |||||
| case FORMAT_U8: | |||||
| m_convert = convert_float_u8; | |||||
| m_codecCtx->sample_fmt = AV_SAMPLE_FMT_U8; | |||||
| break; | |||||
| case FORMAT_S16: | |||||
| m_convert = convert_float_s16; | |||||
| m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S16; | |||||
| break; | |||||
| case FORMAT_S32: | |||||
| m_convert = convert_float_s32; | |||||
| m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S32; | |||||
| break; | |||||
| case FORMAT_FLOAT64: | |||||
| m_convert = convert_float_double; | |||||
| m_codecCtx->sample_fmt = AV_SAMPLE_FMT_DBL; | |||||
| break; | |||||
| default: | |||||
| m_convert = convert_copy<sample_t>; | |||||
| m_codecCtx->sample_fmt = AV_SAMPLE_FMT_FLT; | |||||
| break; | |||||
| } | |||||
| if(m_formatCtx->oformat->flags & AVFMT_GLOBALHEADER) | |||||
| m_codecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER; | |||||
| bool format_supported = false; | |||||
| for(int i = 0; codec->sample_fmts[i] != -1; i++) | |||||
| { | |||||
| if(av_get_alt_sample_fmt(codec->sample_fmts[i], false) == m_codecCtx->sample_fmt) | |||||
| { | |||||
| m_deinterleave = av_sample_fmt_is_planar(codec->sample_fmts[i]); | |||||
| m_codecCtx->sample_fmt = codec->sample_fmts[i]; | |||||
| format_supported = true; | |||||
| } | |||||
| } | |||||
| if(!format_supported) | |||||
| { | |||||
| int chosen_index = 0; | |||||
| auto chosen = av_get_alt_sample_fmt(codec->sample_fmts[chosen_index], false); | |||||
| for(int i = 1; codec->sample_fmts[i] != -1; i++) | |||||
| { | |||||
| auto fmt = av_get_alt_sample_fmt(codec->sample_fmts[i], false); | |||||
| if((fmt > chosen && chosen < m_codecCtx->sample_fmt) || (fmt > m_codecCtx->sample_fmt && fmt < chosen)) | |||||
| { | |||||
| chosen = fmt; | |||||
| chosen_index = i; | |||||
| } | |||||
| } | |||||
| m_codecCtx->sample_fmt = codec->sample_fmts[chosen_index]; | |||||
| m_deinterleave = av_sample_fmt_is_planar(m_codecCtx->sample_fmt); | |||||
| switch(av_get_alt_sample_fmt(m_codecCtx->sample_fmt, false)) | |||||
| { | |||||
| case AV_SAMPLE_FMT_U8: | |||||
| specs.format = FORMAT_U8; | |||||
| m_convert = convert_float_u8; | |||||
| break; | |||||
| case AV_SAMPLE_FMT_S16: | |||||
| specs.format = FORMAT_S16; | |||||
| m_convert = convert_float_s16; | |||||
| break; | |||||
| case AV_SAMPLE_FMT_S32: | |||||
| specs.format = FORMAT_S32; | |||||
| m_convert = convert_float_s32; | |||||
| break; | |||||
| case AV_SAMPLE_FMT_FLT: | |||||
| specs.format = FORMAT_FLOAT32; | |||||
| m_convert = convert_copy<sample_t>; | |||||
| break; | |||||
| case AV_SAMPLE_FMT_DBL: | |||||
| specs.format = FORMAT_FLOAT64; | |||||
| m_convert = convert_float_double; | |||||
| break; | |||||
| default: | |||||
| AUD_THROW(FileException, "File couldn't be written, sample format not supported with ffmpeg."); | |||||
| } | |||||
| } | |||||
| m_codecCtx->sample_rate = 0; | |||||
| if(codec->supported_samplerates) | |||||
| { | |||||
| for(int i = 0; codec->supported_samplerates[i]; i++) | |||||
| { | |||||
| if(codec->supported_samplerates[i] == m_specs.rate) | |||||
| { | |||||
| m_codecCtx->sample_rate = codec->supported_samplerates[i]; | |||||
| break; | |||||
| } | |||||
| else if((codec->supported_samplerates[i] > m_codecCtx->sample_rate && m_specs.rate > m_codecCtx->sample_rate) || | |||||
| (codec->supported_samplerates[i] < m_codecCtx->sample_rate && m_specs.rate < codec->supported_samplerates[i])) | |||||
| { | |||||
| m_codecCtx->sample_rate = codec->supported_samplerates[i]; | |||||
| } | |||||
| } | |||||
| } | |||||
| if(m_codecCtx->sample_rate == 0) | |||||
| m_codecCtx->sample_rate = m_specs.rate; | |||||
| m_specs.rate = m_codecCtx->sample_rate; | |||||
| m_codecCtx->codec_id = m_outputFmt->audio_codec; | |||||
| m_codecCtx->codec_type = AVMEDIA_TYPE_AUDIO; | |||||
| m_codecCtx->bit_rate = bitrate; | |||||
| m_codecCtx->channels = m_specs.channels; | |||||
| m_stream->time_base.num = m_codecCtx->time_base.num = 1; | |||||
| m_stream->time_base.den = m_codecCtx->time_base.den = m_codecCtx->sample_rate; | |||||
| if(avcodec_open2(m_codecCtx, codec, nullptr) < 0) | |||||
| AUD_THROW(FileException, "File couldn't be written, encoder couldn't be opened with ffmpeg."); | |||||
| int samplesize = std::max(int(AUD_SAMPLE_SIZE(m_specs)), AUD_DEVICE_SAMPLE_SIZE(m_specs)); | |||||
| if((m_input_size = m_codecCtx->frame_size)) | |||||
| m_input_buffer.resize(m_input_size * samplesize); | |||||
| if(avio_open(&m_formatCtx->pb, filename.c_str(), AVIO_FLAG_WRITE)) | |||||
| AUD_THROW(FileException, "File couldn't be written, file opening failed with ffmpeg."); | |||||
| avformat_write_header(m_formatCtx, nullptr); | |||||
| } | |||||
| catch(Exception&) | |||||
| { | |||||
| avformat_free_context(m_formatCtx); | |||||
| throw; | |||||
| } | |||||
| } | |||||
| FFMPEGWriter::~FFMPEGWriter() | |||||
| { | |||||
| // writte missing data | |||||
| if(m_input_samples) | |||||
| encode(); | |||||
| close(); | |||||
| av_write_trailer(m_formatCtx); | |||||
| avcodec_close(m_codecCtx); | |||||
| avio_close(m_formatCtx->pb); | |||||
| avformat_free_context(m_formatCtx); | |||||
| } | |||||
| int FFMPEGWriter::getPosition() const | |||||
| { | |||||
| return m_position; | |||||
| } | |||||
| DeviceSpecs FFMPEGWriter::getSpecs() const | |||||
| { | |||||
| return m_specs; | |||||
| } | |||||
| void 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 = std::min(m_input_size - m_input_samples, length); | |||||
| std::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(); | |||||
| m_input_samples = 0; | |||||
| } | |||||
| } | |||||
| } | |||||
| else // PCM data, can write directly! | |||||
| { | |||||
| int samplesize = AUD_SAMPLE_SIZE(m_specs); | |||||
| m_input_buffer.assureSize(length * std::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); | |||||
| m_input_samples = length; | |||||
| encode(); | |||||
| m_position += length; | |||||
| } | |||||
| } | |||||
| AUD_NAMESPACE_END | |||||