背景
ffmpeg 通过 avcodec_alloc_context3 解析编码器,本文主要来解析一下,这个函数主要做了什么。
具体代码分析
主要是创建了 AVCodecContext ,并给结构体参数赋予初值。
初值设置主要分成两块,1. 所有编码器都相同的部分;2.每个编码器独有的参数设置。
接下来主要分析后者是如何设置的。
AVCodecContext *avcodec_alloc_context3(const AVCodec *codec)
{
AVCodecContext *avctx= av_malloc(sizeof(AVCodecContext));
if (!avctx)
return NULL;
if (init_context_defaults(avctx, codec) < 0) {
av_free(avctx);
return NULL;
}
return avctx;
}
static int init_context_defaults(AVCodecContext *s, const AVCodec *codec)
{
const FFCodec *const codec2 = ffcodec(codec);
int flags=0;
memset(s, 0, sizeof(AVCodecContext));
s->av_class = &av_codec_context_class;
s->codec_type = codec ? codec->type : AVMEDIA_TYPE_UNKNOWN;
if (codec) {
s->codec = codec;
s->codec_id = codec->id;
}
if(s->codec_type == AVMEDIA_TYPE_AUDIO)
flags= AV_OPT_FLAG_AUDIO_PARAM;
else if(s->codec_type == AVMEDIA_TYPE_VIDEO)
flags= AV_OPT_FLAG_VIDEO_PARAM;
else if(s->codec_type == AVMEDIA_TYPE_SUBTITLE)
flags= AV_OPT_FLAG_SUBTITLE_PARAM;
av_opt_set_defaults2(s, flags, flags);
av_channel_layout_uninit(&s->ch_layout);
s->time_base = (AVRational){0,1};
s->framerate = (AVRational){ 0, 1 };
s->pkt_timebase = (AVRational){ 0, 1 };
s->get_buffer2 = avcodec_default_get_buffer2;
s->get_format = avcodec_default_get_format;
s->get_encode_buffer = avcodec_default_get_encode_buffer;
s->execute = avcodec_default_execute;
s->execute2 = avcodec_default_execute2;
s->sample_aspect_ratio = (AVRational){0,1};
s->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
s->pix_fmt = AV_PIX_FMT_NONE;
s->sw_pix_fmt = AV_PIX_FMT_NONE;
s->sample_fmt = AV_SAMPLE_FMT_NONE;
s->reordered_opaque = AV_NOPTS_VALUE;
if(codec && codec2->priv_data_size){
s->priv_data = av_mallocz(codec2->priv_data_size);
if (!s->priv_data)
return AVERROR(ENOMEM);
if(codec->priv_class){
*(const AVClass**)s->priv_data = codec->priv_class;
av_opt_set_defaults(s->priv_data);
}
}
if (codec && codec2->defaults) {
int ret;
const FFCodecDefault *d = codec2->defaults;
while (d->key) {
ret = av_opt_set(s, d->key, d->value, 0);
av_assert0(ret >= 0);
d++;
}
}
return 0;
}
每个编码器独有参数设置
avcodec_alloc_context3 时,会将 codec2->priv_data_size 创建大小的内存赋值给 priv_data,并将 codec 的priv_class 赋值给 priv_data。
av_opt_set_defaults 会把 options 中的默认值,设置到 priv_data 中,相当于给编码器的参数结构体设置了默认值。
if(codec && codec2->priv_data_size){
s->priv_data = av_mallocz(codec2->priv_data_size);
if (!s->priv_data)
return AVERROR(ENOMEM);
if(codec->priv_class){
*(const AVClass**)s->priv_data = codec->priv_class;
av_opt_set_defaults(s->priv_data);
}
}
为啥 codec 的priv_class 可以直接赋值给 priv_data。以 aacenc 为例,codec 的 priv_class 为 &aacenc_class,而 codec 的 priv_data_size 为 sizeof(AACEncContext)。所有相当于 AVCodecContext 创建了 sizeof(AACEncContext) 大小的 priv_data。之所以可以这样,是因为 AACEncContext 第一个参数就是 AVClass*。相当于这也是 ffmpeg 的小技巧,应该所有的编码器变量结构体,第一个参数都是 AVClass* (看了一下 nvenc.h 也是这样的)。
static const AVClass aacenc_class = {
.class_name = "AAC encoder",
.item_name = av_default_item_name,
.option = aacenc_options,
.version = LIBAVUTIL_VERSION_INT,
};
const FFCodec ff_aac_encoder = {
.p.name = "aac",
CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_AAC,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
AV_CODEC_CAP_SMALL_LAST_FRAME,
.priv_data_size = sizeof(AACEncContext),
.init = aac_encode_init,
FF_CODEC_ENCODE_CB(aac_encode_frame),
.close = aac_encode_end,
.defaults = aac_encode_defaults,
.p.supported_samplerates = ff_mpeg4audio_sample_rates,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.p.priv_class = &aacenc_class,
};
typedef struct AACEncContext {
AVClass *av_class;
AACEncOptions options; ///< encoding options
PutBitContext pb;
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
AVFloatDSPContext *fdsp;
AACPCEInfo pce; ///< PCE data, if needed
float *planar_samples[16]; ///< saved preprocessed input
int profile; ///< copied from avctx
int needs_pce; ///< flag for non-standard layout
LPCContext lpc; ///< used by TNS
int samplerate_index; ///< MPEG-4 samplerate index
int channels; ///< channel count
const uint8_t *reorder_map; ///< lavc to aac reorder map
const uint8_t *chan_map; ///< channel configuration map
ChannelElement *cpe; ///< channel elements
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
const AACCoefficientsEncoder *coder;
int cur_channel; ///< current channel for coder context
int random_state;
float lambda;
int last_frame_pb_count; ///< number of bits for the previous frame
float lambda_sum; ///< sum(lambda), for Qvg reporting
int lambda_count; ///< count(lambda), for Qvg reporting
enum RawDataBlockType cur_type; ///< channel group type cur_channel belongs to
AudioFrameQueue afq;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
uint16_t quantize_band_cost_cache_generation;
AACQuantizeBandCostCacheEntry quantize_band_cost_cache[256][128]; ///< memoization area for quantize_band_cost
void (*abs_pow34)(float *out, const float *in, const int size);
void (*quant_bands)(int *out, const float *in, const float *scaled,
int size, int is_signed, int maxval, const float Q34,
const float rounding);
struct {
float *samples;
} buffer;
} AACEncContext;