Files
workinf_Blender_Wasm/blender-5.2.0/extern/audaspace/plugins/ffmpeg/FFMPEGWriter.cpp
mes123456 f3e21c6b98
Some checks failed
M6 deployable RC / quick (push) Has been cancelled
M6 deployable RC / chromium (push) Has been cancelled
M6 deployable RC / release (push) Has been cancelled
chore: checkpoint web Blender migration source, tools and docs
Accumulated uncommitted work: GHOST Web backend, WebGL/GPU web layer,
web workflow engine and protocol, tooling, docs and task records.
Large generated evidence, user-testing archives, build snapshots and
imported test assets remain untracked.

Co-Authored-By: Claude Code <noreply@anthropic.com>
2026-11-10 11:18:01 +08:00

642 lines
18 KiB
C++

/*******************************************************************************
* Copyright 2009-2024 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>
#if LIBAVCODEC_VERSION_MAJOR >= 59
#include <libavutil/channel_layout.h>
#endif
}
AUD_NAMESPACE_BEGIN
/* FFmpeg < 4.0 */
#if LIBAVCODEC_VERSION_MAJOR < 58
#define FFMPEG_OLD_CODE
#endif
/* FFmpeg < 5.0 */
#if LIBAVCODEC_VERSION_MAJOR < 59
#define FFMPEG_OLD_CH_LAYOUT
#endif
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 = sizeof(sample_t);
for(int channel = 0; channel < m_specs.channels; channel++)
{
for(int i = 0; i < m_input_samples; 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_convert != convert_copy<sample_t>)
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_samples * AUD_SAMPLE_SIZE(m_specs));
}
else
// convert first
if(m_convert != convert_copy<sample_t>)
m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(data), m_input_samples * m_specs.channels);
#ifdef FFMPEG_OLD_CODE
m_packet->data = nullptr;
m_packet->size = 0;
av_init_packet(m_packet);
av_frame_unref(m_frame);
int got_packet;
#endif
m_frame->nb_samples = m_input_samples;
m_frame->format = m_codecCtx->sample_fmt;
m_frame->sample_rate = m_codecCtx->sample_rate;
#ifdef FFMPEG_OLD_CH_LAYOUT
m_frame->channel_layout = m_codecCtx->channel_layout;
m_frame->channels = m_specs.channels;
#else
if(av_channel_layout_copy(&m_frame->ch_layout, &m_codecCtx->ch_layout) < 0)
AUD_THROW(FileException, "File couldn't be written, couldn't copy audio channel layout.");
#endif
if(avcodec_fill_audio_frame(m_frame, m_specs.channels, m_codecCtx->sample_fmt, reinterpret_cast<data_t*>(data), m_input_buffer.getSize(), 1) < 0)
AUD_THROW(FileException, "File couldn't be written, filling the audio frame failed with ffmpeg.");
AVRational sample_time = { 1, static_cast<int>(m_specs.rate) };
m_frame->pts = av_rescale_q(m_position - m_input_samples, sample_time, m_codecCtx->time_base);
#ifdef FFMPEG_OLD_CODE
if(avcodec_encode_audio2(m_codecCtx, m_packet, m_frame, &got_packet))
{
AUD_THROW(FileException, "File couldn't be written, audio encoding failed with ffmpeg.");
}
if(got_packet)
{
m_packet->flags |= AV_PKT_FLAG_KEY;
m_packet->stream_index = m_stream->index;
if(av_write_frame(m_formatCtx, m_packet) < 0)
{
av_free_packet(m_packet);
AUD_THROW(FileException, "Frame couldn't be writen to the file with ffmpeg.");
}
av_free_packet(m_packet);
}
#else
if(avcodec_send_frame(m_codecCtx, m_frame) < 0)
AUD_THROW(FileException, "File couldn't be written, audio encoding failed with ffmpeg.");
int result;
while((result = avcodec_receive_packet(m_codecCtx, m_packet)) == 0)
{
m_packet->stream_index = m_stream->index;
av_packet_rescale_ts(m_packet, m_codecCtx->time_base, m_stream->time_base);
const int written = av_write_frame(m_formatCtx, m_packet);
av_packet_unref(m_packet);
if(written < 0)
AUD_THROW(FileException, "Frame couldn't be written to the file with ffmpeg.");
}
if(result != AVERROR(EAGAIN) && result != AVERROR_EOF)
AUD_THROW(FileException, "FFmpeg failed to receive an encoded audio packet.");
#endif
}
void FFMPEGWriter::close()
{
#ifdef FFMPEG_OLD_CODE
int got_packet = true;
while(got_packet)
{
m_packet->data = nullptr;
m_packet->size = 0;
av_init_packet(m_packet);
if(avcodec_encode_audio2(m_codecCtx, m_packet, nullptr, &got_packet))
AUD_THROW(FileException, "File end couldn't be written, audio encoding failed with ffmpeg.");
if(got_packet)
{
m_packet->flags |= AV_PKT_FLAG_KEY;
m_packet->stream_index = m_stream->index;
if(av_write_frame(m_formatCtx, m_packet))
{
av_free_packet(m_packet);
AUD_THROW(FileException, "Final frames couldn't be writen to the file with ffmpeg.");
}
av_free_packet(m_packet);
}
}
#else
if(avcodec_send_frame(m_codecCtx, nullptr) < 0)
AUD_THROW(FileException, "File couldn't be written, audio encoding failed with ffmpeg.");
int result;
while((result = avcodec_receive_packet(m_codecCtx, m_packet)) == 0)
{
m_packet->stream_index = m_stream->index;
av_packet_rescale_ts(m_packet, m_codecCtx->time_base, m_stream->time_base);
const int written = av_write_frame(m_formatCtx, m_packet);
av_packet_unref(m_packet);
if(written < 0)
AUD_THROW(FileException, "Frame couldn't be written to the file with ffmpeg.");
}
if(result != AVERROR(EAGAIN) && result != AVERROR_EOF)
AUD_THROW(FileException, "FFmpeg failed to receive an encoded audio packet.");
#endif
}
FFMPEGWriter::FFMPEGWriter(const std::string &filename, DeviceSpecs specs, Container format, Codec codec, unsigned int bitrate) :
m_position(0),
m_specs(specs),
m_formatCtx(nullptr),
m_codecCtx(nullptr),
m_stream(nullptr),
m_packet(nullptr),
m_frame(nullptr),
m_input_samples(0),
m_deinterleave(false),
m_finished(false),
m_failed(false)
{
static const char* formats[] = { nullptr, "ac3", "flac", "matroska", "mp2", "mp3", "ogg", "wav", "adts" };
if(format <= CONTAINER_INVALID || format > CONTAINER_AAC)
AUD_THROW(FileException, "Invalid audio container.");
if((format == CONTAINER_AAC && codec != CODEC_AAC) ||
(format == CONTAINER_AC3 && codec != CODEC_AC3) ||
(format == CONTAINER_MP2 && codec != CODEC_MP2))
AUD_THROW(FileException, "The audio codec does not match the selected container.");
try
{
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.");
const AVOutputFormat* outputFmt = m_formatCtx->oformat;
if(!outputFmt) {
AUD_THROW(FileException, "File couldn't be written, output format couldn't be found with ffmpeg.");
}
AVCodecID audio_codec = AV_CODEC_ID_NONE;
switch(codec)
{
case CODEC_AAC:
audio_codec = AV_CODEC_ID_AAC;
break;
case CODEC_AC3:
audio_codec = AV_CODEC_ID_AC3;
break;
case CODEC_FLAC:
audio_codec = AV_CODEC_ID_FLAC;
break;
case CODEC_MP2:
audio_codec = AV_CODEC_ID_MP2;
break;
case CODEC_MP3:
audio_codec = AV_CODEC_ID_MP3;
break;
case CODEC_OPUS:
audio_codec = AV_CODEC_ID_OPUS;
break;
case CODEC_PCM:
switch(specs.format)
{
case FORMAT_U8:
audio_codec = AV_CODEC_ID_PCM_U8;
break;
case FORMAT_S16:
audio_codec = AV_CODEC_ID_PCM_S16LE;
break;
case FORMAT_S24:
audio_codec = AV_CODEC_ID_PCM_S24LE;
break;
case FORMAT_S32:
audio_codec = AV_CODEC_ID_PCM_S32LE;
break;
case FORMAT_FLOAT32:
audio_codec = AV_CODEC_ID_PCM_F32LE;
break;
case FORMAT_FLOAT64:
audio_codec = AV_CODEC_ID_PCM_F64LE;
break;
default:
audio_codec = AV_CODEC_ID_NONE;
break;
}
break;
case CODEC_VORBIS:
audio_codec = AV_CODEC_ID_VORBIS;
break;
default:
audio_codec = AV_CODEC_ID_NONE;
break;
}
uint64_t channel_layout = 0;
switch(m_specs.channels)
{
case CHANNELS_MONO:
channel_layout = AV_CH_LAYOUT_MONO;
break;
case CHANNELS_STEREO:
channel_layout = AV_CH_LAYOUT_STEREO;
break;
case CHANNELS_STEREO_LFE:
channel_layout = AV_CH_LAYOUT_2POINT1;
break;
case CHANNELS_SURROUND4:
channel_layout = AV_CH_LAYOUT_QUAD;
break;
case CHANNELS_SURROUND5:
channel_layout = AV_CH_LAYOUT_5POINT0_BACK;
break;
case CHANNELS_SURROUND51:
channel_layout = AV_CH_LAYOUT_5POINT1_BACK;
break;
case CHANNELS_SURROUND61:
channel_layout = AV_CH_LAYOUT_6POINT1_BACK;
break;
case CHANNELS_SURROUND71:
channel_layout = AV_CH_LAYOUT_7POINT1;
break;
default:
AUD_THROW(FileException, "File couldn't be written, channel layout not supported.");
}
if(audio_codec == AV_CODEC_ID_NONE)
AUD_THROW(FileException, "File couldn't be written, audio codec not found with ffmpeg.");
const AVCodec* codec = avcodec_find_encoder(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;
#ifdef FFMPEG_OLD_CODE
m_codecCtx = m_stream->codec;
#else
m_codecCtx = avcodec_alloc_context3(codec);
#endif
if(!m_codecCtx)
AUD_THROW(FileException, "File couldn't be written, context creation failed with ffmpeg.");
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 |= AV_CODEC_FLAG_GLOBAL_HEADER;
const AVSampleFormat* sample_formats = nullptr;
const int* sample_rates = nullptr;
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(61, 19, 100)
const void* configurations = nullptr;
if(avcodec_get_supported_config(m_codecCtx, codec, AV_CODEC_CONFIG_SAMPLE_FORMAT,
0, &configurations, nullptr) < 0)
AUD_THROW(FileException, "Couldn't query FFmpeg sample formats.");
sample_formats = static_cast<const AVSampleFormat*>(configurations);
if(avcodec_get_supported_config(m_codecCtx, codec, AV_CODEC_CONFIG_SAMPLE_RATE,
0, &configurations, nullptr) < 0)
AUD_THROW(FileException, "Couldn't query FFmpeg sample rates.");
sample_rates = static_cast<const int*>(configurations);
#else
sample_formats = codec->sample_fmts;
sample_rates = codec->supported_samplerates;
#endif
bool format_supported = sample_formats == nullptr;
for(int i = 0; sample_formats && sample_formats[i] != -1; i++)
{
if(av_get_alt_sample_fmt(sample_formats[i], false) == m_codecCtx->sample_fmt)
{
m_deinterleave = av_sample_fmt_is_planar(sample_formats[i]);
m_codecCtx->sample_fmt = sample_formats[i];
format_supported = true;
}
}
if(!format_supported)
{
int chosen_index = 0;
auto chosen = av_get_alt_sample_fmt(sample_formats[chosen_index], false);
for(int i = 1; sample_formats[i] != -1; i++)
{
auto fmt = av_get_alt_sample_fmt(sample_formats[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 = sample_formats[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_specs.format = specs.format;
m_codecCtx->sample_rate = 0;
if(sample_rates)
{
for(int i = 0; sample_rates[i]; i++)
{
if(sample_rates[i] == m_specs.rate)
{
m_codecCtx->sample_rate = sample_rates[i];
break;
}
else if((sample_rates[i] > m_codecCtx->sample_rate && m_specs.rate > m_codecCtx->sample_rate) ||
(sample_rates[i] < m_codecCtx->sample_rate && m_specs.rate < sample_rates[i]))
{
m_codecCtx->sample_rate = sample_rates[i];
}
}
}
if(m_codecCtx->sample_rate == 0)
m_codecCtx->sample_rate = m_specs.rate;
#ifdef __EMSCRIPTEN__
if(m_specs.rate != m_codecCtx->sample_rate)
AUD_THROW(FileException, "The selected sample rate is not supported by this audio encoder.");
#endif
m_specs.rate = m_codecCtx->sample_rate;
#ifdef FFMPEG_OLD_CODE
m_codecCtx->codec_id = audio_codec;
#endif
m_codecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
m_codecCtx->bit_rate = bitrate;
#ifdef FFMPEG_OLD_CH_LAYOUT
m_codecCtx->channel_layout = channel_layout;
m_codecCtx->channels = m_specs.channels;
#else
av_channel_layout_uninit(&m_codecCtx->ch_layout);
av_channel_layout_from_mask(&m_codecCtx->ch_layout, channel_layout);
#endif
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.");
#ifndef FFMPEG_OLD_CODE
if(avcodec_parameters_from_context(m_stream->codecpar, m_codecCtx) < 0)
AUD_THROW(FileException, "File couldn't be written, codec parameters couldn't be copied to the context.");
#endif
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.");
if(avformat_write_header(m_formatCtx, nullptr) < 0)
AUD_THROW(FileException, "File couldn't be written, writing the header failed.");
#ifdef FFMPEG_OLD_CODE
m_packet = new AVPacket({});
#else
m_packet = av_packet_alloc();
#endif
m_frame = av_frame_alloc();
if(!m_packet || !m_frame)
AUD_THROW(FileException, "Cannot allocate FFmpeg audio frame/packet.");
}
catch(...)
{
release();
throw;
}
}
void FFMPEGWriter::release() noexcept
{
av_frame_free(&m_frame);
#ifdef FFMPEG_OLD_CODE
delete m_packet;
m_packet = nullptr;
if(m_codecCtx)
avcodec_close(m_codecCtx);
#else
av_packet_free(&m_packet);
avcodec_free_context(&m_codecCtx);
#endif
if(m_formatCtx)
{
avio_closep(&m_formatCtx->pb);
avformat_free_context(m_formatCtx);
m_formatCtx = nullptr;
}
}
void FFMPEGWriter::finish()
{
if(m_failed)
AUD_THROW(FileException, "FFmpeg audio output previously failed.");
if(m_finished)
return;
try
{
if(m_input_samples)
{
// Fixed-size encoders without small-last-frame support need zero padding.
if(m_input_size && m_input_samples < m_input_size &&
!(m_codecCtx->codec->capabilities & AV_CODEC_CAP_SMALL_LAST_FRAME))
{
std::memset(m_input_buffer.getBuffer() + m_input_samples * m_specs.channels, 0,
(m_input_size - m_input_samples) * AUD_SAMPLE_SIZE(m_specs));
m_position += m_input_size - m_input_samples;
m_input_samples = m_input_size;
}
encode();
m_input_samples = 0;
}
close();
if(av_write_trailer(m_formatCtx) < 0)
AUD_THROW(FileException, "FFmpeg could not write the audio trailer.");
avio_flush(m_formatCtx->pb);
const int io_error = m_formatCtx->pb->error;
const int close_error = avio_closep(&m_formatCtx->pb);
if(io_error < 0 || close_error < 0)
AUD_THROW(FileException, "FFmpeg could not finalize the audio file.");
m_finished = true;
release();
}
catch(...)
{
m_failed = true;
release();
throw;
}
}
FFMPEGWriter::~FFMPEGWriter()
{
// Existing desktop callers still finalize on destruction. Web mixdown calls
// finish explicitly while failure can abort the staged-file transaction.
if(!m_finished && !m_failed)
{
try { finish(); }
catch(...) {}
}
release();
}
int FFMPEGWriter::getPosition() const
{
return m_position;
}
DeviceSpecs FFMPEGWriter::getSpecs() const
{
return m_specs;
}
void FFMPEGWriter::write(unsigned int length, sample_t* buffer)
{
if(m_finished || m_failed)
AUD_THROW(FileException, "Cannot write to closed FFmpeg audio output.");
try
{
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();
std::memcpy(buf, buffer, length * samplesize);
m_input_samples = length;
m_position += length;
encode();
m_input_samples = 0;
}
}
catch(...)
{
m_failed = true;
release();
throw;
}
}
AUD_NAMESPACE_END