static void noise_gate_update(void *data, obs_data_t *s)
{
struct noise_gate_data *ng = data;
float open_threshold_db;
float close_threshold_db;
float sample_rate;
int attack_time_ms;
int hold_time_ms;
int release_time_ms;
open_threshold_db = (float)obs_data_get_double(s, S_OPEN_THRESHOLD);
close_threshold_db = (float)obs_data_get_double(s, S_CLOSE_THRESHOLD);
attack_time_ms = (int)obs_data_get_int(s, S_ATTACK_TIME);
hold_time_ms = (int)obs_data_get_int(s, S_HOLD_TIME);
release_time_ms = (int)obs_data_get_int(s, S_RELEASE_TIME);
sample_rate = (float)audio_output_get_sample_rate(obs_get_audio());
ng->sample_rate_i = 1.0f / sample_rate;
ng->channels = audio_output_get_channels(obs_get_audio());
ng->open_threshold = db_to_mul(open_threshold_db);
ng->close_threshold = db_to_mul(close_threshold_db);
ng->attack_rate = 1.0f / (ms_to_secf(attack_time_ms) * sample_rate);
ng->release_rate = 1.0f / (ms_to_secf(release_time_ms) * sample_rate);
const float threshold_diff = ng->open_threshold - ng->close_threshold;
const float min_decay_period = (1.0f / 75.0f) * sample_rate;
ng->decay_rate = threshold_diff / min_decay_period;
ng->hold_time = ms_to_secf(hold_time_ms);
ng->is_open = false;
ng->attenuation = 0.0f;
ng->level = 0.0f;
ng->held_time = 0.0f;
}
static void expander_update(void *data, obs_data_t *s)
{
struct expander_data *cd = data;
const char *presets = obs_data_get_string(s, S_PRESETS);
if (strcmp(presets, "expander") == 0 && cd->is_gate) {
obs_data_clear(s);
obs_data_set_string(s, S_PRESETS, "expander");
expander_defaults(s);
cd->is_gate = false;
}
if (strcmp(presets, "gate") == 0 && !cd->is_gate) {
obs_data_clear(s);
obs_data_set_string(s, S_PRESETS, "gate");
expander_defaults(s);
cd->is_gate = true;
}
const uint32_t sample_rate =
audio_output_get_sample_rate(obs_get_audio());
const size_t num_channels =
audio_output_get_channels(obs_get_audio());
const float attack_time_ms =
(float)obs_data_get_int(s, S_ATTACK_TIME);
const float release_time_ms =
(float)obs_data_get_int(s, S_RELEASE_TIME);
const float output_gain_db =
(float)obs_data_get_double(s, S_OUTPUT_GAIN);
cd->ratio = (float)obs_data_get_double(s, S_RATIO);
cd->threshold = (float)obs_data_get_double(s, S_THRESHOLD);
cd->attack_gain = gain_coefficient(sample_rate,
attack_time_ms / MS_IN_S_F);
cd->release_gain = gain_coefficient(sample_rate,
release_time_ms / MS_IN_S_F);
cd->output_gain = db_to_mul(output_gain_db);
cd->num_channels = num_channels;
cd->sample_rate = sample_rate;
cd->slope = 1.0f - cd->ratio;
const char *detect_mode = obs_data_get_string(s, S_DETECTOR);
if (strcmp(detect_mode, "RMS") == 0)
cd->detector = RMS_DETECT;
if (strcmp(detect_mode, "peak") == 0)
cd->detector = PEAK_DETECT;
if (strcmp(detect_mode, "none") == 0)
cd->detector = NO_DETECT;
size_t sample_len = sample_rate * DEFAULT_AUDIO_BUF_MS / MS_IN_S;
if (cd->envelope_buf_len == 0)
resize_env_buffer(cd, sample_len);
if (cd->runaverage_len == 0)
resize_runaverage_buffer(cd, sample_len);
if (cd->maxspl_len == 0)
resize_maxspl_buffer(cd, sample_len);
if (cd->env_in_len == 0)
resize_env_in_buffer(cd, sample_len);
}
static void add_audio_encoder_params(struct dstr *cmd, obs_encoder_t *aencoder)
{
obs_data_t *settings = obs_encoder_get_settings(aencoder);
int bitrate = (int)obs_data_get_int(settings, "bitrate");
audio_t *audio = obs_get_audio();
struct dstr name = {0};
obs_data_release(settings);
dstr_copy(&name, obs_encoder_get_name(aencoder));
dstr_replace(&name, "\"", "\"\"");
dstr_catf(cmd, "\"%s\" %d %d %d ",
name.array,
bitrate,
(int)obs_encoder_get_sample_rate(aencoder),
(int)audio_output_get_channels(audio));
dstr_free(&name);
}
static void *aac_create(obs_data_t *settings, obs_encoder_t *encoder)
{
struct aac_encoder *enc;
int bitrate = (int)obs_data_get_int(settings, "bitrate");
audio_t *audio = obs_encoder_audio(encoder);
avcodec_register_all();
enc = bzalloc(sizeof(struct aac_encoder));
enc->encoder = encoder;
enc->aac = avcodec_find_encoder(AV_CODEC_ID_AAC);
blog(LOG_INFO, "---------------------------------");
if (!enc->aac) {
warn("Couldn't find encoder");
goto fail;
}
if (!bitrate) {
warn("Invalid bitrate specified");
return NULL;
}
enc->context = avcodec_alloc_context3(enc->aac);
if (!enc->context) {
warn("Failed to create codec context");
goto fail;
}
enc->context->bit_rate = bitrate * 1000;
enc->context->channels = (int)audio_output_get_channels(audio);
enc->context->sample_rate = audio_output_get_sample_rate(audio);
enc->context->sample_fmt = enc->aac->sample_fmts ?
enc->aac->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
/* if using FFmpeg's AAC encoder, at least set a cutoff value
* (recommended by konverter) */
if (strcmp(enc->aac->name, "aac") == 0) {
int cutoff1 = 4000 + (int)enc->context->bit_rate / 8;
int cutoff2 = 12000 + (int)enc->context->bit_rate / 8;
int cutoff3 = enc->context->sample_rate / 2;
int cutoff;
cutoff = MIN(cutoff1, cutoff2);
cutoff = MIN(cutoff, cutoff3);
enc->context->cutoff = cutoff;
}
info("bitrate: %" PRId64 ", channels: %d",
enc->context->bit_rate / 1000, enc->context->channels);
init_sizes(enc, audio);
/* enable experimental FFmpeg encoder if the only one available */
enc->context->strict_std_compliance = -2;
enc->context->flags = CODEC_FLAG_GLOBAL_HEADER;
if (initialize_codec(enc))
return enc;
fail:
aac_destroy(enc);
return NULL;
}
static void *libfdk_create(obs_data_t settings, obs_encoder_t encoder)
{
bool hasFdkHandle = false;
libfdk_encoder_t *enc = 0;
int bitrate = (int)obs_data_get_int(settings, "bitrate") * 1000;
int afterburner = obs_data_get_bool(settings, "afterburner") ? 1 : 0;
audio_t audio = obs_encoder_audio(encoder);
int mode = 0;
AACENC_ERROR err;
if (!bitrate) {
blog(LOG_ERROR, "Invalid bitrate");
return NULL;
}
enc = bzalloc(sizeof(libfdk_encoder_t));
enc->encoder = encoder;
enc->channels = (int)audio_output_get_channels(audio);
enc->sample_rate = audio_output_get_sample_rate(audio);
switch(enc->channels) {
case 1:
mode = MODE_1;
break;
case 2:
mode = MODE_2;
break;
case 3:
mode = MODE_1_2;
break;
case 4:
mode = MODE_1_2_1;
break;
case 5:
mode = MODE_1_2_2;
break;
case 6:
mode = MODE_1_2_2_1;
break;
default:
blog(LOG_ERROR, "Invalid channel count");
goto fail;
}
CHECK_LIBFDK(aacEncOpen(&enc->fdkhandle, 0, enc->channels));
hasFdkHandle = true;
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_AOT,
2)); // MPEG-4 AAC-LC
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_SAMPLERATE,
enc->sample_rate));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_CHANNELMODE, mode));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_CHANNELORDER, 1));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_BITRATEMODE, 0));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_BITRATE, bitrate));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_TRANSMUX, 0));
CHECK_LIBFDK(aacEncoder_SetParam(enc->fdkhandle, AACENC_AFTERBURNER,
afterburner));
CHECK_LIBFDK(aacEncEncode(enc->fdkhandle, NULL, NULL, NULL, NULL));
CHECK_LIBFDK(aacEncInfo(enc->fdkhandle, &enc->info));
enc->frame_size_bytes = enc->info.frameLength * 2 * enc->channels;
enc->packet_buffer_size = enc->channels * 768;
if(enc->packet_buffer_size < 8192)
enc->packet_buffer_size = 8192;
enc->packet_buffer = bmalloc(enc->packet_buffer_size);
blog(LOG_INFO, "libfdk_aac encoder created");
blog(LOG_INFO, "libfdk_aac bitrate: %d, channels: %d",
bitrate / 1000, enc->channels);
return enc;
fail:
if(hasFdkHandle)
aacEncClose(&enc->fdkhandle);
if(enc->packet_buffer)
bfree(enc->packet_buffer);
if(enc)
bfree(enc);
blog(LOG_WARNING, "libfdk_aac encoder creation failed");
return 0;
}
static bool build_flv_meta_data(obs_output_t *context,
uint8_t **output, size_t *size, size_t a_idx)
{
obs_encoder_t *vencoder = obs_output_get_video_encoder(context);
obs_encoder_t *aencoder = obs_output_get_audio_encoder(context, a_idx);
video_t *video = obs_encoder_video(vencoder);
audio_t *audio = obs_encoder_audio(aencoder);
char buf[4096];
char *enc = buf;
char *end = enc+sizeof(buf);
struct dstr encoder_name = {0};
if (a_idx > 0 && !aencoder)
return false;
enc_str(&enc, end, "onMetaData");
*enc++ = AMF_ECMA_ARRAY;
enc = AMF_EncodeInt32(enc, end, a_idx == 0 ? 14 : 9);
enc_num_val(&enc, end, "duration", 0.0);
enc_num_val(&enc, end, "fileSize", 0.0);
if (a_idx == 0) {
enc_num_val(&enc, end, "width",
(double)obs_encoder_get_width(vencoder));
enc_num_val(&enc, end, "height",
(double)obs_encoder_get_height(vencoder));
enc_str_val(&enc, end, "videocodecid", "avc1");
enc_num_val(&enc, end, "videodatarate",
encoder_bitrate(vencoder));
enc_num_val(&enc, end, "framerate",
video_output_get_frame_rate(video));
}
enc_str_val(&enc, end, "audiocodecid", "mp4a");
enc_num_val(&enc, end, "audiodatarate", encoder_bitrate(aencoder));
enc_num_val(&enc, end, "audiosamplerate",
(double)obs_encoder_get_sample_rate(aencoder));
enc_num_val(&enc, end, "audiosamplesize", 16.0);
enc_num_val(&enc, end, "audiochannels",
(double)audio_output_get_channels(audio));
enc_bool_val(&enc, end, "stereo",
audio_output_get_channels(audio) == 2);
enc_bool_val(&enc, end, "2.1",
audio_output_get_channels(audio) == 3);
enc_bool_val(&enc, end, "3.1",
audio_output_get_channels(audio) == 4);
enc_bool_val(&enc, end, "4.0",
audio_output_get_channels(audio) == 4);
enc_bool_val(&enc, end, "4.1",
audio_output_get_channels(audio) == 5);
enc_bool_val(&enc, end, "5.1",
audio_output_get_channels(audio) == 6);
enc_bool_val(&enc, end, "7.1",
audio_output_get_channels(audio) == 8);
dstr_printf(&encoder_name, "%s (libobs version ",
MODULE_NAME);
#ifdef HAVE_OBSCONFIG_H
dstr_cat(&encoder_name, OBS_VERSION);
#else
dstr_catf(&encoder_name, "%d.%d.%d",
LIBOBS_API_MAJOR_VER,
LIBOBS_API_MINOR_VER,
LIBOBS_API_PATCH_VER);
#endif
dstr_cat(&encoder_name, ")");
enc_str_val(&enc, end, "encoder", encoder_name.array);
dstr_free(&encoder_name);
*enc++ = 0;
*enc++ = 0;
*enc++ = AMF_OBJECT_END;
*size = enc-buf;
*output = bmemdup(buf, *size);
return true;
}
static void *enc_create(obs_data_t *settings, obs_encoder_t *encoder,
const char *type, const char *alt)
{
struct enc_encoder *enc;
int bitrate = (int)obs_data_get_int(settings, "bitrate");
audio_t *audio = obs_encoder_audio(encoder);
avcodec_register_all();
enc = bzalloc(sizeof(struct enc_encoder));
enc->encoder = encoder;
enc->codec = avcodec_find_encoder_by_name(type);
enc->type = type;
if (!enc->codec && alt) {
enc->codec = avcodec_find_encoder_by_name(alt);
enc->type = alt;
}
blog(LOG_INFO, "---------------------------------");
if (!enc->codec) {
warn("Couldn't find encoder");
goto fail;
}
if (!bitrate) {
warn("Invalid bitrate specified");
return NULL;
}
enc->context = avcodec_alloc_context3(enc->codec);
if (!enc->context) {
warn("Failed to create codec context");
goto fail;
}
enc->context->bit_rate = bitrate * 1000;
enc->context->channels = (int)audio_output_get_channels(audio);
enc->context->sample_rate = audio_output_get_sample_rate(audio);
enc->context->sample_fmt = enc->codec->sample_fmts ?
enc->codec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
/* check to make sure sample rate is supported */
if (enc->codec->supported_samplerates) {
const int *rate = enc->codec->supported_samplerates;
int cur_rate = enc->context->sample_rate;
int closest = 0;
while (*rate) {
int dist = abs(cur_rate - *rate);
int closest_dist = abs(cur_rate - closest);
if (dist < closest_dist)
closest = *rate;
rate++;
}
if (closest)
enc->context->sample_rate = closest;
}
/* if using FFmpeg's AAC encoder, at least set a cutoff value
* (recommended by konverter) */
if (strcmp(enc->codec->name, "aac") == 0) {
int cutoff1 = 4000 + (int)enc->context->bit_rate / 8;
int cutoff2 = 12000 + (int)enc->context->bit_rate / 8;
int cutoff3 = enc->context->sample_rate / 2;
int cutoff;
cutoff = MIN(cutoff1, cutoff2);
cutoff = MIN(cutoff, cutoff3);
enc->context->cutoff = cutoff;
}
info("bitrate: %" PRId64 ", channels: %d",
enc->context->bit_rate / 1000, enc->context->channels);
init_sizes(enc, audio);
/* enable experimental FFmpeg encoder if the only one available */
enc->context->strict_std_compliance = -2;
enc->context->flags = CODEC_FLAG_GLOBAL_HEADER;
if (initialize_codec(enc))
return enc;
fail:
enc_destroy(enc);
return NULL;
}
bool audio_callback(void *param,
uint64_t start_ts_in, uint64_t end_ts_in, uint64_t *out_ts,
uint32_t mixers, struct audio_output_data *mixes)
{
struct obs_core_data *data = &obs->data;
struct obs_core_audio *audio = &obs->audio;
struct obs_source *source;
size_t sample_rate = audio_output_get_sample_rate(audio->audio);
size_t channels = audio_output_get_channels(audio->audio);
struct ts_info ts = {start_ts_in, end_ts_in};
size_t audio_size;
uint64_t min_ts;
da_resize(audio->render_order, 0);
da_resize(audio->root_nodes, 0);
circlebuf_push_back(&audio->buffered_timestamps, &ts, sizeof(ts));
circlebuf_peek_front(&audio->buffered_timestamps, &ts, sizeof(ts));
min_ts = ts.start;
audio_size = AUDIO_OUTPUT_FRAMES * sizeof(float);
#if DEBUG_AUDIO == 1
blog(LOG_DEBUG, "ts %llu-%llu", ts.start, ts.end);
#endif
/* ------------------------------------------------ */
/* build audio render order
* NOTE: these are source channels, not audio channels */
for (uint32_t i = 0; i < MAX_CHANNELS; i++) {
obs_source_t *source = obs_get_output_source(i);
if (source) {
obs_source_enum_active_tree(source, push_audio_tree,
audio);
push_audio_tree(NULL, source, audio);
da_push_back(audio->root_nodes, &source);
obs_source_release(source);
}
}
pthread_mutex_lock(&data->audio_sources_mutex);
source = data->first_audio_source;
while (source) {
push_audio_tree(NULL, source, audio);
source = (struct obs_source*)source->next_audio_source;
}
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* render audio data */
for (size_t i = 0; i < audio->render_order.num; i++) {
obs_source_t *source = audio->render_order.array[i];
obs_source_audio_render(source, mixers, channels, sample_rate,
audio_size);
}
/* ------------------------------------------------ */
/* get minimum audio timestamp */
pthread_mutex_lock(&data->audio_sources_mutex);
calc_min_ts(data, sample_rate, &min_ts);
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* if a source has gone backward in time, buffer */
if (min_ts < ts.start)
add_audio_buffering(audio, sample_rate, &ts, min_ts);
/* ------------------------------------------------ */
/* mix audio */
if (!audio->buffering_wait_ticks) {
for (size_t i = 0; i < audio->root_nodes.num; i++) {
obs_source_t *source = audio->root_nodes.array[i];
if (source->audio_pending)
continue;
pthread_mutex_lock(&source->audio_buf_mutex);
if (source->audio_output_buf[0][0] && source->audio_ts)
mix_audio(mixes, source, channels, sample_rate,
&ts);
pthread_mutex_unlock(&source->audio_buf_mutex);
}
}
/* ------------------------------------------------ */
/* discard audio */
pthread_mutex_lock(&data->audio_sources_mutex);
source = data->first_audio_source;
while (source) {
pthread_mutex_lock(&source->audio_buf_mutex);
discard_audio(audio, source, channels, sample_rate, &ts);
pthread_mutex_unlock(&source->audio_buf_mutex);
source = (struct obs_source*)source->next_audio_source;
}
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* release audio sources */
release_audio_sources(audio);
circlebuf_pop_front(&audio->buffered_timestamps, NULL, sizeof(ts));
*out_ts = ts.start;
if (audio->buffering_wait_ticks) {
audio->buffering_wait_ticks--;
return false;
}
UNUSED_PARAMETER(param);
return true;
}
