Add FFmpeg's AAC enoder
This just adds FFmpeg's default AAC encoder as an audio encoder. Going to try to start getting things going with the RTMP output library next.master
jp9000
parent
8c74db9ffc
commit
cabe98cb4e
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@ -20,11 +20,15 @@ find_package(Libswresample REQUIRED)
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include_directories(${Libswresample_INCLUDE_DIR})
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add_definitions(${Libswresample_DEFINITIONS})
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set(obs-ffmpeg_HEADERS
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obs-ffmpeg-formats.h)
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set(obs-ffmpeg_SOURCES
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obs-ffmpeg.c
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obs-ffmpeg-aac.c
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obs-ffmpeg-output.c)
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add_library(obs-ffmpeg MODULE
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${obs-ffmpeg_HEADERS}
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${obs-ffmpeg_SOURCES})
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target_link_libraries(obs-ffmpeg
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libobs
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@ -0,0 +1,269 @@
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/******************************************************************************
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Copyright (C) 2014 by Hugh Bailey <obs.jim@gmail.com>
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This program is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program. If not, see <http://www.gnu.org/licenses/>.
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******************************************************************************/
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#include <util/base.h>
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#include <util/circlebuf.h>
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#include <obs.h>
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#include <libavformat/avformat.h>
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#include "obs-ffmpeg-formats.h"
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struct aac_encoder {
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obs_encoder_t encoder;
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AVCodec *aac;
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AVCodecContext *context;
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struct circlebuf buffers[MAX_AV_PLANES];
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uint8_t *samples[MAX_AV_PLANES];
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AVFrame *aframe;
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int total_samples;
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size_t audio_planes;
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size_t audio_size;
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int frame_size; /* pretty much always 1024 for AAC */
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int frame_size_bytes;
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};
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static const char *aac_getname(const char *locale)
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{
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UNUSED_PARAMETER(locale);
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return "FFmpeg Default AAC Encoder";
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}
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static void aac_warn(const char *func, const char *format, ...)
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{
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va_list args;
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char msg[1024];
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va_start(args, format);
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vsnprintf(msg, sizeof(msg), format, args);
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blog(LOG_WARNING, "[%s}: %s", func, msg);
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va_end(args);
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}
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static void aac_destroy(void *data)
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{
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struct aac_encoder *enc = data;
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for (size_t i = 0; i < MAX_AV_PLANES; i++)
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circlebuf_free(&enc->buffers[i]);
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if (enc->samples[0])
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av_freep(&enc->samples[0]);
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if (enc->context)
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avcodec_close(enc->context);
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if (enc->aframe)
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av_frame_free(&enc->aframe);
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bfree(enc);
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}
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static bool initialize_codec(struct aac_encoder *enc)
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{
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int ret;
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enc->aframe = av_frame_alloc();
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if (!enc->aframe) {
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aac_warn("initialize_codec", "Failed to allocate audio frame");
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return false;
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}
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ret = avcodec_open2(enc->context, enc->aac, NULL);
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if (ret < 0) {
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aac_warn("initialize_codec", "Failed to open AAC codec: %s",
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av_err2str(ret));
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return false;
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}
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enc->frame_size = enc->context->frame_size;
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if (!enc->frame_size)
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enc->frame_size = 1024;
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enc->frame_size_bytes = enc->frame_size * (int)enc->audio_size;
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ret = av_samples_alloc(enc->samples, NULL, enc->context->channels,
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enc->frame_size, enc->context->sample_fmt, 0);
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if (ret < 0) {
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aac_warn("initialize_codec", "Failed to create audio buffer: "
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"%s", av_err2str(ret));
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return false;
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}
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return true;
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}
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static void *aac_create(obs_data_t settings, obs_encoder_t encoder)
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{
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struct aac_encoder *enc;
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int bitrate = (int)obs_data_getint(settings, "bitrate");
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audio_t audio = obs_encoder_audio(encoder);
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if (!bitrate) {
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aac_warn("aac_create", "Invalid bitrate specified");
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return NULL;
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}
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enc = bzalloc(sizeof(struct aac_encoder));
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enc->encoder = encoder;
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enc->aac = avcodec_find_encoder(AV_CODEC_ID_AAC);
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if (!enc->aac) {
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aac_warn("aac_create", "Couldn't find encoder");
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goto fail;
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}
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avcodec_register(enc->aac);
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enc->context = avcodec_alloc_context3(enc->aac);
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if (!enc->context) {
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aac_warn("aac_create", "Failed to create codec context");
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goto fail;
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}
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enc->context->bit_rate = bitrate;
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enc->context->channels = (int)audio_output_channels(audio);
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enc->context->sample_rate = audio_output_samplerate(audio);
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enc->context->sample_fmt = enc->aac->sample_fmts ?
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enc->aac->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
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enc->audio_planes = audio_output_planes(audio);
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enc->audio_size = audio_output_blocksize(audio);
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/* enable experimental FFmpeg encoder if the only one available */
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enc->context->strict_std_compliance = -2;
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if (initialize_codec(enc))
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return enc;
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fail:
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aac_destroy(enc);
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return NULL;
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}
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static bool do_aac_encode(struct aac_encoder *enc,
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struct encoder_packet *packet, bool *received_packet)
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{
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AVRational time_base = {1, enc->context->sample_rate};
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AVPacket avpacket = {0};
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int got_packet;
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int ret;
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enc->aframe->nb_samples = enc->frame_size;
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enc->aframe->pts = av_rescale_q(enc->total_samples,
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(AVRational){1, enc->context->sample_rate},
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enc->context->time_base);
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ret = avcodec_fill_audio_frame(enc->aframe, enc->context->channels,
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enc->context->sample_fmt, enc->samples[0],
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enc->frame_size_bytes * enc->context->channels, 1);
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if (ret < 0) {
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aac_warn("do_aac_encode", "avcodec_fill_audio_frame failed: %s",
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av_err2str(ret));
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return false;
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}
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enc->total_samples += enc->frame_size;
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ret = avcodec_encode_audio2(enc->context, &avpacket, enc->aframe,
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&got_packet);
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if (ret < 0) {
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aac_warn("do_aac_encode", "avcodec_encode_audio2 failed: %s",
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av_err2str(ret));
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return false;
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}
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*received_packet = !!got_packet;
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if (!got_packet)
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return true;
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packet->pts = rescale_ts(avpacket.pts, enc->context, time_base);
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packet->dts = rescale_ts(avpacket.dts, enc->context, time_base);
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packet->data = bmemdup(avpacket.data, avpacket.size);
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packet->type = OBS_ENCODER_AUDIO;
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packet->timebase_num = 1;
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packet->timebase_den = (int32_t)enc->context->sample_rate;
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av_free_packet(&avpacket);
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return true;
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}
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static bool aac_encode(void *data, struct encoder_frame *frame,
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struct encoder_packet *packet, bool *received_packet)
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{
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struct aac_encoder *enc = data;
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size_t size = frame->frames * enc->audio_size;
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for (size_t i = 0; i < enc->audio_planes; i++)
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circlebuf_push_back(&enc->buffers[i], frame->data[i], size);
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if (enc->buffers[0].size < (size_t)enc->frame_size_bytes) {
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*received_packet = false;
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return true;
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}
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for (size_t i = 0; i < enc->audio_planes; i++)
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circlebuf_pop_front(&enc->buffers[i], enc->samples[i],
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enc->frame_size_bytes);
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return do_aac_encode(enc, packet, received_packet);
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}
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static void aac_defaults(obs_data_t settings)
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{
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obs_data_setint(settings, "bitrate", 128);
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}
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static obs_properties_t aac_properties(const char *locale)
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{
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obs_properties_t props = obs_properties_create(locale);
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/* TODO: locale */
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obs_properties_add_int(props, "bitrate", "Bitrate",
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32, 320, 32);
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return props;
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}
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static bool aac_extra_data(void *data, uint8_t **extra_data, size_t *size)
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{
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struct aac_encoder *enc = data;
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*extra_data = enc->context->extradata;
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*size = enc->context->extradata_size;
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return true;
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}
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static bool aac_audio_info(void *data, struct audio_convert_info *info)
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{
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struct aac_encoder *enc = data;
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memset(info, 0, sizeof(struct audio_convert_info));
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info->format = convert_ffmpeg_sample_format(enc->context->sample_fmt);
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return true;
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}
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struct obs_encoder_info aac_encoder_info = {
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.id = "ffmpeg_aac",
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.type = OBS_ENCODER_AUDIO,
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.codec = "AAC",
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.getname = aac_getname,
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.create = aac_create,
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.destroy = aac_destroy,
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.encode = aac_encode,
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.defaults = aac_defaults,
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.properties = aac_properties,
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.extra_data = aac_extra_data,
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.audio_info = aac_audio_info
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};
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@ -1,5 +1,12 @@
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#pragma once
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static inline int64_t rescale_ts(int64_t val, AVCodecContext *context,
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AVRational new_base)
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{
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return av_rescale_q_rnd(val, context->time_base, new_base,
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AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX);
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}
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static inline enum AVPixelFormat obs_to_ffmpeg_video_format(
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enum video_format format)
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{
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@ -454,14 +454,6 @@ static void ffmpeg_output_destroy(void *data)
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}
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}
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static inline int64_t rescale_ts(int64_t val, AVCodecContext *context,
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AVStream *stream)
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{
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return av_rescale_q_rnd(val, context->time_base,
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stream->time_base,
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AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX);
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}
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static inline void copy_data(AVPicture *pic, const struct video_data *frame,
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int height)
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{
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@ -532,9 +524,9 @@ static void receive_video(void *param, struct video_data *frame)
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if (!ret && got_packet && packet.size) {
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packet.pts = rescale_ts(packet.pts, context,
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data->video);
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data->video->time_base);
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packet.dts = rescale_ts(packet.dts, context,
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data->video);
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data->video->time_base);
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packet.duration = (int)av_rescale_q(packet.duration,
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context->time_base,
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data->video->time_base);
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@ -592,8 +584,8 @@ static void encode_audio(struct ffmpeg_output *output,
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if (!got_packet)
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return;
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packet.pts = rescale_ts(packet.pts, context, data->audio);
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packet.dts = rescale_ts(packet.dts, context, data->audio);
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packet.pts = rescale_ts(packet.pts, context, data->audio->time_base);
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packet.dts = rescale_ts(packet.dts, context, data->audio->time_base);
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packet.duration = (int)av_rescale_q(packet.duration, context->time_base,
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data->audio->time_base);
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packet.stream_index = data->audio->index;
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@ -171,9 +171,13 @@
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</PostBuildEvent>
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</ItemDefinitionGroup>
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<ItemGroup>
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<ClCompile Include="..\..\..\plugins\obs-ffmpeg\obs-ffmpeg-aac.c" />
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<ClCompile Include="..\..\..\plugins\obs-ffmpeg\obs-ffmpeg-output.c" />
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<ClCompile Include="..\..\..\plugins\obs-ffmpeg\obs-ffmpeg.c" />
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</ItemGroup>
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<ItemGroup>
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<ClInclude Include="..\..\..\plugins\obs-ffmpeg\obs-ffmpeg-formats.h" />
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</ItemGroup>
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<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
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<ImportGroup Label="ExtensionTargets">
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</ImportGroup>
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@ -21,5 +21,13 @@
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<ClCompile Include="..\..\..\plugins\obs-ffmpeg\obs-ffmpeg-output.c">
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<Filter>Source Files</Filter>
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</ClCompile>
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<ClCompile Include="..\..\..\plugins\obs-ffmpeg\obs-ffmpeg-aac.c">
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<Filter>Source Files</Filter>
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</ClCompile>
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</ItemGroup>
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<ItemGroup>
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<ClInclude Include="..\..\..\plugins\obs-ffmpeg\obs-ffmpeg-formats.h">
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<Filter>Header Files</Filter>
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</ClInclude>
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</ItemGroup>
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</Project>
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