New upstream version 21.0.2+dfsg1
This commit is contained in:
Sebastian Ramacher
parent
1f1bbb3518
commit
baafb6325b
706 changed files with 49633 additions and 5044 deletions
411
plugins/obs-ffmpeg/obs-ffmpeg-audio-encoders.c
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411
plugins/obs-ffmpeg/obs-ffmpeg-audio-encoders.c
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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 <util/darray.h>
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#include <obs-module.h>
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#include <libavformat/avformat.h>
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#include "obs-ffmpeg-formats.h"
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#include "obs-ffmpeg-compat.h"
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#define do_log(level, format, ...) \
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blog(level, "[FFmpeg %s encoder: '%s'] " format, \
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enc->type, \
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obs_encoder_get_name(enc->encoder), \
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##__VA_ARGS__)
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#define warn(format, ...) do_log(LOG_WARNING, format, ##__VA_ARGS__)
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#define info(format, ...) do_log(LOG_INFO, format, ##__VA_ARGS__)
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#define debug(format, ...) do_log(LOG_DEBUG, format, ##__VA_ARGS__)
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struct enc_encoder {
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obs_encoder_t *encoder;
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const char *type;
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AVCodec *codec;
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AVCodecContext *context;
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uint8_t *samples[MAX_AV_PLANES];
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AVFrame *aframe;
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int64_t total_samples;
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DARRAY(uint8_t) packet_buffer;
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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 inline uint64_t convert_speaker_layout(enum speaker_layout layout)
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{
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switch (layout) {
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case SPEAKERS_UNKNOWN: return 0;
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case SPEAKERS_MONO: return AV_CH_LAYOUT_MONO;
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case SPEAKERS_STEREO: return AV_CH_LAYOUT_STEREO;
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case SPEAKERS_2POINT1: return AV_CH_LAYOUT_SURROUND;
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case SPEAKERS_4POINT0: return AV_CH_LAYOUT_4POINT0;
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case SPEAKERS_4POINT1: return AV_CH_LAYOUT_4POINT1;
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case SPEAKERS_5POINT1: return AV_CH_LAYOUT_5POINT1_BACK;
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case SPEAKERS_7POINT1: return AV_CH_LAYOUT_7POINT1;
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}
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/* shouldn't get here */
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return 0;
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}
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static inline enum speaker_layout convert_ff_channel_layout(uint64_t channel_layout)
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{
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switch (channel_layout) {
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case AV_CH_LAYOUT_MONO: return SPEAKERS_MONO;
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case AV_CH_LAYOUT_STEREO: return SPEAKERS_STEREO;
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case AV_CH_LAYOUT_SURROUND: return SPEAKERS_2POINT1;
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case AV_CH_LAYOUT_4POINT0: return SPEAKERS_4POINT0;
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case AV_CH_LAYOUT_4POINT1: return SPEAKERS_4POINT1;
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case AV_CH_LAYOUT_5POINT1_BACK: return SPEAKERS_5POINT1;
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case AV_CH_LAYOUT_7POINT1: return SPEAKERS_7POINT1;
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}
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/* shouldn't get here */
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return SPEAKERS_UNKNOWN;
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}
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static const char *aac_getname(void *unused)
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{
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UNUSED_PARAMETER(unused);
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return obs_module_text("FFmpegAAC");
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}
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static const char *opus_getname(void *unused)
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{
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UNUSED_PARAMETER(unused);
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return obs_module_text("FFmpegOpus");
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}
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static void enc_destroy(void *data)
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{
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struct enc_encoder *enc = data;
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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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da_free(enc->packet_buffer);
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bfree(enc);
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}
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static bool initialize_codec(struct enc_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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warn("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->codec, NULL);
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if (ret < 0) {
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warn("Failed to open AAC codec: %s", 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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warn("Failed to create audio buffer: %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 init_sizes(struct enc_encoder *enc, audio_t *audio)
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{
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const struct audio_output_info *aoi;
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enum audio_format format;
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aoi = audio_output_get_info(audio);
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format = convert_ffmpeg_sample_format(enc->context->sample_fmt);
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enc->audio_planes = get_audio_planes(format, aoi->speakers);
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enc->audio_size = get_audio_size(format, aoi->speakers, 1);
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}
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#ifndef MIN
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#define MIN(x, y) ((x) < (y) ? (x) : (y))
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#endif
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static void *enc_create(obs_data_t *settings, obs_encoder_t *encoder,
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const char *type, const char *alt)
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{
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struct enc_encoder *enc;
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int bitrate = (int)obs_data_get_int(settings, "bitrate");
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audio_t *audio = obs_encoder_audio(encoder);
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avcodec_register_all();
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enc = bzalloc(sizeof(struct enc_encoder));
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enc->encoder = encoder;
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enc->codec = avcodec_find_encoder_by_name(type);
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enc->type = type;
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if (!enc->codec && alt) {
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enc->codec = avcodec_find_encoder_by_name(alt);
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enc->type = alt;
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}
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blog(LOG_INFO, "---------------------------------");
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if (!enc->codec) {
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warn("Couldn't find encoder");
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goto fail;
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}
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if (!bitrate) {
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warn("Invalid bitrate specified");
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return NULL;
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}
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enc->context = avcodec_alloc_context3(enc->codec);
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if (!enc->context) {
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warn("Failed to create codec context");
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goto fail;
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}
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enc->context->bit_rate = bitrate * 1000;
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const struct audio_output_info *aoi;
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aoi = audio_output_get_info(audio);
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enc->context->channels = (int)audio_output_get_channels(audio);
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enc->context->channel_layout = convert_speaker_layout(aoi->speakers);
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enc->context->sample_rate = audio_output_get_sample_rate(audio);
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enc->context->sample_fmt = enc->codec->sample_fmts ?
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enc->codec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
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/* check to make sure sample rate is supported */
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if (enc->codec->supported_samplerates) {
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const int *rate = enc->codec->supported_samplerates;
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int cur_rate = enc->context->sample_rate;
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int closest = 0;
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while (*rate) {
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int dist = abs(cur_rate - *rate);
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int closest_dist = abs(cur_rate - closest);
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if (dist < closest_dist)
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closest = *rate;
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rate++;
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}
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if (closest)
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enc->context->sample_rate = closest;
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}
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/* if using FFmpeg's AAC encoder, at least set a cutoff value
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* (recommended by konverter) */
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if (strcmp(enc->codec->name, "aac") == 0) {
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int cutoff1 = 4000 + (int)enc->context->bit_rate / 8;
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int cutoff2 = 12000 + (int)enc->context->bit_rate / 8;
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int cutoff3 = enc->context->sample_rate / 2;
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int cutoff;
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cutoff = MIN(cutoff1, cutoff2);
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cutoff = MIN(cutoff, cutoff3);
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enc->context->cutoff = cutoff;
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}
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info("bitrate: %" PRId64 ", channels: %d, channel_layout: %x\n",
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(int64_t)enc->context->bit_rate / 1000,
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(int)enc->context->channels,
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(unsigned int)enc->context->channel_layout);
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init_sizes(enc, 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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enc->context->flags = CODEC_FLAG_GLOBAL_H;
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if (initialize_codec(enc))
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return enc;
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fail:
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enc_destroy(enc);
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return NULL;
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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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return enc_create(settings, encoder, "aac", NULL);
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}
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static void *opus_create(obs_data_t *settings, obs_encoder_t *encoder)
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{
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return enc_create(settings, encoder, "libopus", "opus");
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}
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static bool do_encode(struct enc_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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warn("avcodec_fill_audio_frame failed: %s", 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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#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(57, 40, 101)
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ret = avcodec_send_frame(enc->context, enc->aframe);
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if (ret == 0)
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ret = avcodec_receive_packet(enc->context, &avpacket);
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got_packet = (ret == 0);
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if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
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ret = 0;
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#else
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ret = avcodec_encode_audio2(enc->context, &avpacket, enc->aframe,
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&got_packet);
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#endif
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if (ret < 0) {
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warn("avcodec_encode_audio2 failed: %s", 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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da_resize(enc->packet_buffer, 0);
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da_push_back_array(enc->packet_buffer, avpacket.data, avpacket.size);
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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 = enc->packet_buffer.array;
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packet->size = 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 enc_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 enc_encoder *enc = data;
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for (size_t i = 0; i < enc->audio_planes; i++)
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memcpy(enc->samples[i], frame->data[i], enc->frame_size_bytes);
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return do_encode(enc, packet, received_packet);
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}
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static void enc_defaults(obs_data_t *settings)
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{
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obs_data_set_default_int(settings, "bitrate", 128);
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}
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static obs_properties_t *enc_properties(void *unused)
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{
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UNUSED_PARAMETER(unused);
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obs_properties_t *props = obs_properties_create();
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obs_properties_add_int(props, "bitrate",
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obs_module_text("Bitrate"), 64, 1024, 32);
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return props;
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}
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static bool enc_extra_data(void *data, uint8_t **extra_data, size_t *size)
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{
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struct enc_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 void enc_audio_info(void *data, struct audio_convert_info *info)
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{
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struct enc_encoder *enc = data;
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info->format = convert_ffmpeg_sample_format(enc->context->sample_fmt);
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info->samples_per_sec = (uint32_t)enc->context->sample_rate;
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info->speakers = convert_ff_channel_layout(enc->context->channel_layout);
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}
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static size_t enc_frame_size(void *data)
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{
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struct enc_encoder *enc =data;
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return enc->frame_size;
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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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.get_name = aac_getname,
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.create = aac_create,
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.destroy = enc_destroy,
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.encode = enc_encode,
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.get_frame_size = enc_frame_size,
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.get_defaults = enc_defaults,
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.get_properties = enc_properties,
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.get_extra_data = enc_extra_data,
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.get_audio_info = enc_audio_info
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};
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struct obs_encoder_info opus_encoder_info = {
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.id = "ffmpeg_opus",
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.type = OBS_ENCODER_AUDIO,
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.codec = "opus",
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.get_name = opus_getname,
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.create = opus_create,
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.destroy = enc_destroy,
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.encode = enc_encode,
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.get_frame_size = enc_frame_size,
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.get_defaults = enc_defaults,
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.get_properties = enc_properties,
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.get_extra_data = enc_extra_data,
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.get_audio_info = enc_audio_info
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};
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