FFmpeg3.2在msvc msys环境下源码编译，如何一步步改写为长尾？

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材料+VS2019+FFmpeg3.2源码+GitHub+ksvc/FFmpeg: 镜像于git:source.ffmpeg.org/ffmpeg.git，支持RTMP协议扩展的H.265/HEVC，由KSYUN提供。x264（需使用msvc msys源码编译）+备注：最新版x264需修改。

材料

VS2019

FFmpeg3.2源码

GitHub - ksvc/FFmpeg: mirror of git://source.ffmpeg.org/ffmpeg.git, with RTMP protocol extensions for H.265/HEVC powered by KSYUN.

x264 （要求采用msvc+msys 源码编译）

备注：最新版本x264需要修改FFmpeg源码libavcodec/libx264.c中x264_bit_depth为X264_BIT_DEPTH

x265（要求采用msvc+msys 源码编译）

fdk-aac（要求采用msvc+msys 源码编译）

备注：最新版本fdk-aac需要按照下面问题，进行FFmpeg源码libavcodec/libfdk-aacenc.c修改

注意：由于FFmpeg源码的版本太久，采用的第三方库是最新的，因此需要做调整

基本操作

编译64位FFmpeg程序

Windows开始菜单 -> Visual Studio 2019 -> x64 Native Tools Command Prompt for VS 2019

编译32位FFmpeg程序

Windows开始菜单 -> Visual Studio 2019 -> x86 Native Tools Command Prompt for VS 2019

作用：使用VS的开发环境变量，调用cl.exe等VS开发工具集

打开msys

msys2_shell.cmd -defterm -full-path -no-start -here -mingw32

mingw32说明编译的是32位版本

关键点

1设置正确的链接器（指定MSVC的链接器）

我们使用的是微软的编译器cl.exe和链接器link.exe，然而msys2自带有link.exe,和msvc 的link.exe重名，且前者所在目录在环境变量中靠前，所以运行link命令时实际运行的是msys2的link.exe,这将造成链接出错,按照如下操作修改名称，从而调用msvc 的link.exe

# whereis cl

cl: /d/vs2019/IDE/VC/Tools/MSVC/14.29.30133/bin/HostX86/x86/cl.exe

# whereis link

link: /usr/bin/link.exe /d/vs2019/IDE/VC/Tools/MSVC/14.29.30133/bin/HostX86/x86/link.exe /usr/share/man/man1/link.1.gz

mv /usr/bin/link.exe /usr/bin/msyslink.exe

#whereis link

link: /d/vs2019/IDE/VC/Tools/MSVC/14.29.30133/bin/HostX86/x86/link.exe /usr/share/man/man1/link.1.gz

2将x264 x265等库文件的安装路径文件pkg添加到环境变量

export PKG_CONFIG_PATH=$PKG_CONFIG_PATH:/usr/local/lib/pkgconfig/

编译指令

./configure --enable-shared --prefix=/home/out --toolchain=msvc --cc=cl --cxx=cl --enable-libx264 --enable-libx265 --enable-gpl --enable-libfdk-aac --enable-nonfree --extra-cflags=-I/usr/local/include --extra-ldflags=-LIBPATH:/usr/local/lib

./configure --enable-shared --prefix=/home/out --toolchain=msvc --cc=cl --cxx=cl --enable-libx264 --enable-libx265 --enable-gpl --enable-libfdk-aac --enable-nonfree --extra-cflags=-I/usr/local/include --extra-ldflags=-LIBPATH:/usr/local/lib

-LIBPATH是微软编译器链接时，指定的关键字，跟GCC -L是同样的效果，但GCC -L不能被微软编译器链接识别到，切记！

--toolchain=msvc 指定使用微软编译器编译

--enable-gpl 链接x264 x265需要同意该协议

--enable-nonfree 链接fdk-aac需要同意该协议

--extra-cflags指定x264,x265等第三方库的头文件目录

--extra-ldflags指定x264,x265等第三方库的LIB文件目录

FFmpeg默认是动态链接其他的库，如何静态链接暂时不清楚

静态库编译出来的是.a文件，修改名称.lib就可以使用

问题

ERROR: libx264 not found

libx264.lib找不到，这是因为生成的x264库默认命名为libx264.dll.lib，将其改为libx264.lib可解决这个问题

ERROR: libfdk_aac not found

fdk-aac/aacenc_lib.h: No such file or directory

ERROR: x265 not found using pkg-config

解决方案

export PKG_CONFIG_PATH=$PKG_CONFIG_PATH:/usr/local/lib/pkgconfig/

将libx265.lib 改名为x265.lib后配置成功

--enable-static没有编译出lib文件

只有.a文件

"encoderDelay": 不是 "AACENC_InfoStruct" 的成员

修改源码libavcodec/libfdk-aacenc.c

/* * AAC encoder wrapper * Copyright (c) 2012 Martin Storsjo * * This file is part of FFmpeg. * * Permission to use, copy, modify, and/or distribute this software for any * purpose with or without fee is hereby granted, provided that the above * copyright notice and this permission notice appear in all copies. * * THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES * WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF * MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR * ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES * WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN * ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF * OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. */ #include <fdk-aac/aacenc_lib.h> #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "libavutil/opt.h" #include "avcodec.h" #include "audio_frame_queue.h" #include "internal.h" #define FDKENC_VER_AT_LEAST(vl0, vl1) \ (defined(AACENCODER_LIB_VL0) && \ ((AACENCODER_LIB_VL0 > vl0) || \ (AACENCODER_LIB_VL0 == vl0 && AACENCODER_LIB_VL1 >= vl1))) typedef struct AACContext { const AVClass *class; HANDLE_AACENCODER handle; int afterburner; int eld_sbr; int signaling; int latm; int header_period; int vbr; AudioFrameQueue afq; } AACContext; static const AVOption aac_enc_options[] = { { "afterburner", "Afterburner (improved quality)", offsetof(AACContext, afterburner), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, { "eld_sbr", "Enable SBR for ELD (for SBR in other configurations, use the -profile parameter)", offsetof(AACContext, eld_sbr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, { "signaling", "SBR/PS signaling style", offsetof(AACContext, signaling), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 2, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" }, { "default", "Choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)", 0, AV_OPT_TYPE_CONST, { .i64 = -1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" }, { "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" }, { "explicit_sbr", "Explicit SBR, implicit PS signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" }, { "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" }, { "latm", "Output LATM/LOAS encapsulated data", offsetof(AACContext, latm), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, { "header_period", "StreamMuxConfig and PCE repetition period (in frames)", offsetof(AACContext, header_period), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 0xffff, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, { "vbr", "VBR mode (1-5)", offsetof(AACContext, vbr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 5, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, { NULL } }; static const AVClass aac_enc_class = { "libfdk_aac", av_default_item_name, aac_enc_options, LIBAVUTIL_VERSION_INT }; static const char *aac_get_error(AACENC_ERROR err) { switch (err) { case AACENC_OK: return "No error"; case AACENC_INVALID_HANDLE: return "Invalid handle"; case AACENC_MEMORY_ERROR: return "Memory allocation error"; case AACENC_UNSUPPORTED_PARAMETER: return "Unsupported parameter"; case AACENC_INVALID_CONFIG: return "Invalid config"; case AACENC_INIT_ERROR: return "Initialization error"; case AACENC_INIT_AAC_ERROR: return "AAC library initialization error"; case AACENC_INIT_SBR_ERROR: return "SBR library initialization error"; case AACENC_INIT_TP_ERROR: return "Transport library initialization error"; case AACENC_INIT_META_ERROR: return "Metadata library initialization error"; case AACENC_ENCODE_ERROR: return "Encoding error"; case AACENC_ENCODE_EOF: return "End of file"; default: return "Unknown error"; } } static int aac_encode_close(AVCodecContext *avctx) { AACContext *s = avctx->priv_data; if (s->handle) aacEncClose(&s->handle); av_freep(&avctx->extradata); ff_af_queue_close(&s->afq); return 0; } static av_cold int aac_encode_init(AVCodecContext *avctx) { AACContext *s = avctx->priv_data; int ret = AVERROR(EINVAL); AACENC_InfoStruct info = { 0 }; CHANNEL_MODE mode; AACENC_ERROR err; int aot = FF_PROFILE_AAC_LOW + 1; int sce = 0, cpe = 0; if ((err = aacEncOpen(&s->handle, 0, avctx->channels)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to open the encoder: %s\n", aac_get_error(err)); goto error; } if (avctx->profile != FF_PROFILE_UNKNOWN) aot = avctx->profile + 1; if ((err = aacEncoder_SetParam(s->handle, AACENC_AOT, aot)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set the AOT %d: %s\n", aot, aac_get_error(err)); goto error; } if (aot == FF_PROFILE_AAC_ELD + 1 && s->eld_sbr) { if ((err = aacEncoder_SetParam(s->handle, AACENC_SBR_MODE, 1)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to enable SBR for ELD: %s\n", aac_get_error(err)); goto error; } } if ((err = aacEncoder_SetParam(s->handle, AACENC_SAMPLERATE, avctx->sample_rate)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set the sample rate %d: %s\n", avctx->sample_rate, aac_get_error(err)); goto error; } switch (avctx->channels) { case 1: mode = MODE_1; sce = 1; cpe = 0; break; case 2: mode = MODE_2; sce = 0; cpe = 1; break; case 3: mode = MODE_1_2; sce = 1; cpe = 1; break; case 4: mode = MODE_1_2_1; sce = 2; cpe = 1; break; case 5: mode = MODE_1_2_2; sce = 1; cpe = 2; break; case 6: mode = MODE_1_2_2_1; sce = 2; cpe = 2; break; /* The version macro is introduced the same time as the 7.1 support, so this should suffice. */ #ifdef AACENCODER_LIB_VL0 case 8: sce = 2; cpe = 3; if (avctx->channel_layout == AV_CH_LAYOUT_7POINT1) { mode = MODE_7_1_REAR_SURROUND; } else { // MODE_1_2_2_2_1 and MODE_7_1_FRONT_CENTER use the same channel layout mode = MODE_7_1_FRONT_CENTER; } break; #endif default: av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels %d\n", avctx->channels); goto error; } if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELMODE, mode)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set channel mode %d: %s\n", mode, aac_get_error(err)); goto error; } if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELORDER, 1)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set wav channel order %d: %s\n", mode, aac_get_error(err)); goto error; } if (avctx->flags & AV_CODEC_FLAG_QSCALE || s->vbr) { int mode = s->vbr ? s->vbr : avctx->global_quality; if (mode < 1 || mode > 5) { av_log(avctx, AV_LOG_WARNING, "VBR quality %d out of range, should be 1-5\n", mode); mode = av_clip(mode, 1, 5); } av_log(avctx, AV_LOG_WARNING, "Note, the VBR setting is unsupported and only works with " "some parameter combinations\n"); if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATEMODE, mode)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set the VBR bitrate mode %d: %s\n", mode, aac_get_error(err)); goto error; } } else { if (avctx->bit_rate <= 0) { if (avctx->profile == FF_PROFILE_AAC_HE_V2) { sce = 1; cpe = 0; } avctx->bit_rate = (96*sce + 128*cpe) * avctx->sample_rate / 44; if (avctx->profile == FF_PROFILE_AAC_HE || avctx->profile == FF_PROFILE_AAC_HE_V2 || avctx->profile == FF_PROFILE_MPEG2_AAC_HE || s->eld_sbr) avctx->bit_rate /= 2; } if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATE, avctx->bit_rate)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set the bitrate %"PRId64": %s\n", (int64_t)avctx->bit_rate, aac_get_error(err)); goto error; } } /* Choose bitstream format - if global header is requested, use * raw access units, otherwise use ADTS. */ if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX, avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER ? 0 : s->latm ? 10 : 2)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n", aac_get_error(err)); goto error; } if (s->latm && s->header_period) { if ((err = aacEncoder_SetParam(s->handle, AACENC_HEADER_PERIOD, s->header_period)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set header period: %s\n", aac_get_error(err)); goto error; } } /* If no signaling mode is chosen, use explicit hierarchical signaling * if using mp4 mode (raw access units, with global header) and * implicit signaling if using ADTS. */ if (s->signaling < 0) s->signaling = avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER ? 2 : 0; if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE, s->signaling)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set signaling mode %d: %s\n", s->signaling, aac_get_error(err)); goto error; } if ((err = aacEncoder_SetParam(s->handle, AACENC_AFTERBURNER, s->afterburner)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set afterburner to %d: %s\n", s->afterburner, aac_get_error(err)); goto error; } if (avctx->cutoff > 0) { if (avctx->cutoff < (avctx->sample_rate + 255) >> 8 || avctx->cutoff > 20000) { av_log(avctx, AV_LOG_ERROR, "cutoff valid range is %d-20000\n", (avctx->sample_rate + 255) >> 8); goto error; } if ((err = aacEncoder_SetParam(s->handle, AACENC_BANDWIDTH, avctx->cutoff)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwidth to %d: %s\n", avctx->cutoff, aac_get_error(err)); goto error; } } if ((err = aacEncEncode(s->handle, NULL, NULL, NULL, NULL)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to initialize the encoder: %s\n", aac_get_error(err)); return AVERROR(EINVAL); } if ((err = aacEncInfo(s->handle, &info)) != AACENC_OK) { av_log(avctx, AV_LOG_ERROR, "Unable to get encoder info: %s\n", aac_get_error(err)); goto error; } avctx->frame_size = info.frameLength; //#if FDKENC_VER_AT_LEAST(4, 0) avctx->initial_padding = info.nDelay; //#else //avctx->initial_padding = info.encoderDelay; //#endif ff_af_queue_init(avctx, &s->afq); if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) { avctx->extradata_size = info.confSize; avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE); if (!avctx->extradata) { ret = AVERROR(ENOMEM); goto error; } memcpy(avctx->extradata, info.confBuf, info.confSize); } return 0; error: aac_encode_close(avctx); return ret; } static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { AACContext *s = avctx->priv_data; AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; AACENC_InArgs in_args = { 0 }; AACENC_OutArgs out_args = { 0 }; int in_buffer_identifier = IN_AUDIO_DATA; int in_buffer_size, in_buffer_element_size; int out_buffer_identifier = OUT_BITSTREAM_DATA; int out_buffer_size, out_buffer_element_size; void *in_ptr, *out_ptr; int ret; uint8_t dummy_buf[1]; AACENC_ERROR err; /* handle end-of-stream small frame and flushing */ if (!frame) { /* Must be a non-null pointer, even if it's a dummy. We could use * the address of anything else on the stack as well. */ in_ptr = dummy_buf; in_buffer_size = 0; in_args.numInSamples = -1; } else { in_ptr = frame->data[0]; in_buffer_size = 2 * avctx->channels * frame->nb_samples; in_args.numInSamples = avctx->channels * frame->nb_samples; /* add current frame to the queue */ if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) return ret; } in_buffer_element_size = 2; in_buf.numBufs = 1; in_buf.bufs = &in_ptr; in_buf.bufferIdentifiers = &in_buffer_identifier; in_buf.bufSizes = &in_buffer_size; in_buf.bufElSizes = &in_buffer_element_size; /* The maximum packet size is 6144 bits aka 768 bytes per channel. */ if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels), 0)) < 0) return ret; out_ptr = avpkt->data; out_buffer_size = avpkt->size; out_buffer_element_size = 1; out_buf.numBufs = 1; out_buf.bufs = &out_ptr; out_buf.bufferIdentifiers = &out_buffer_identifier; out_buf.bufSizes = &out_buffer_size; out_buf.bufElSizes = &out_buffer_element_size; if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) { if (!frame && err == AACENC_ENCODE_EOF) return 0; av_log(avctx, AV_LOG_ERROR, "Unable to encode frame: %s\n", aac_get_error(err)); return AVERROR(EINVAL); } if (!out_args.numOutBytes) return 0; /* Get the next frame pts & duration */ ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, &avpkt->duration); avpkt->size = out_args.numOutBytes; *got_packet_ptr = 1; return 0; } static const AVProfile profiles[] = { { FF_PROFILE_AAC_LOW, "LC" }, { FF_PROFILE_AAC_HE, "HE-AAC" }, { FF_PROFILE_AAC_HE_V2, "HE-AACv2" }, { FF_PROFILE_AAC_LD, "LD" }, { FF_PROFILE_AAC_ELD, "ELD" }, { FF_PROFILE_UNKNOWN }, }; static const AVCodecDefault aac_encode_defaults[] = { { "b", "0" }, { NULL } }; static const uint64_t aac_channel_layout[] = { AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_SURROUND, AV_CH_LAYOUT_4POINT0, AV_CH_LAYOUT_5POINT0_BACK, AV_CH_LAYOUT_5POINT1_BACK, #ifdef AACENCODER_LIB_VL0 AV_CH_LAYOUT_7POINT1_WIDE_BACK, AV_CH_LAYOUT_7POINT1, #endif 0, }; static const int aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 0 }; AVCodec ff_libfdk_aac_encoder = { .name = "libfdk_aac", .long_name = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_AAC, .priv_data_size = sizeof(AACContext), .init = aac_encode_init, .encode2 = aac_encode_frame, .close = aac_encode_close, .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, .priv_class = &aac_enc_class, .defaults = aac_encode_defaults, .profiles = profiles, .supported_samplerates = aac_sample_rates, .channel_layouts = aac_channel_layout, };

libx264 is gpl and --enable-gpl is not specified.

ERROR: libx264 not found

error: 'x264_bit_depth' undeclared (first use in this function)

直接修改libavcodec/libx264.c中x264_bit_depth为X264_BIT_DEPTH

无法打开包括文件: “strings.h”

\SDL2\SDL_stdinc.h(63): fatal error C1083: 无法打开包括文件: “strings.h”: No such file or directory

暂时不打算编译ffplay