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
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