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#include <algorithm>
#include <gst/gst.h>
#include <gst/audio/audio.h>
#if defined(WEBRTC0)
#include <webrtc/modules/audio_processing/include/audio_processing.h>
#include <webrtc/modules/interface/module_common_types.h>
#include <webrtc/system_wrappers/include/trace.h>
#elif defined(WEBRTC1) || defined(WEBRTC2)
#include <modules/audio_processing/include/audio_processing.h>
#else
#error "Need to define WEBRTC0, WEBRTC1 or WEBRTC2"
#endif
#define SAMPLE_RATE 48000
#define SAMPLE_CHANNELS 1
struct _DinoPluginsRtpVoiceProcessorNative {
#if defined(WEBRTC0)
webrtc::AudioProcessing *apm;
#elif defined(WEBRTC1) || defined(WEBRTC2)
rtc::scoped_refptr<webrtc::AudioProcessing> apm;
#endif
gint stream_delay = 0;
gint last_median = 0;
gint last_poor_delays = 0;
};
extern "C" void *dino_plugins_rtp_adjust_to_running_time(GstBaseTransform *transform, GstBuffer *buffer) {
GstBuffer *copy = gst_buffer_copy(buffer);
GST_BUFFER_PTS(copy) = gst_segment_to_running_time(&transform->segment, GST_FORMAT_TIME, GST_BUFFER_PTS(buffer));
return copy;
}
extern "C" void *dino_plugins_rtp_voice_processor_init_native(gint stream_delay) {
auto *native = new _DinoPluginsRtpVoiceProcessorNative();
webrtc::Config config;
#ifdef WEBRTC0
config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, 85));
native->apm = webrtc::AudioProcessing::Create(config);
#endif
native->stream_delay = stream_delay;
return native;
}
extern "C" void dino_plugins_rtp_voice_processor_setup_native(void *native_ptr) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
#if defined(WEBRTC0)
webrtc::AudioProcessing *apm = native->apm;
webrtc::ProcessingConfig pconfig;
pconfig.streams[webrtc::ProcessingConfig::kInputStream] =
webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
pconfig.streams[webrtc::ProcessingConfig::kOutputStream] =
webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
pconfig.streams[webrtc::ProcessingConfig::kReverseInputStream] =
webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
pconfig.streams[webrtc::ProcessingConfig::kReverseOutputStream] =
webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
apm->Initialize(pconfig);
apm->high_pass_filter()->Enable(true);
apm->echo_cancellation()->enable_drift_compensation(false);
apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kModerateSuppression);
apm->echo_cancellation()->enable_delay_logging(true);
apm->echo_cancellation()->Enable(true);
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kModerate);
apm->noise_suppression()->Enable(true);
apm->gain_control()->set_analog_level_limits(0, 255);
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
apm->gain_control()->set_target_level_dbfs(3);
apm->gain_control()->set_compression_gain_db(9);
apm->gain_control()->enable_limiter(true);
apm->gain_control()->Enable(true);
apm->voice_detection()->set_likelihood(webrtc::VoiceDetection::Likelihood::kLowLikelihood);
apm->voice_detection()->Enable(true);
#elif defined(WEBRTC1) || defined(WEBRTC2)
webrtc::AudioProcessing::Config config;
rtc::scoped_refptr<webrtc::AudioProcessing> apm = webrtc::AudioProcessingBuilder().Create();
native->apm = apm;
config.high_pass_filter.enabled = true;
config.echo_canceller.enabled = true;
config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::Level::kModerate;
config.noise_suppression.enabled = true;
config.gain_controller1.analog_level_minimum = 0;
config.gain_controller1.analog_level_maximum = 255;
config.gain_controller1.target_level_dbfs = 3;
config.gain_controller1.compression_gain_db = 9;
config.gain_controller1.enable_limiter = true;
config.gain_controller1.enabled = true;
#ifdef WEBRTC1
config.level_estimation.enabled = true;
config.voice_detection.enabled = true;
#endif
apm->ApplyConfig(config);
#endif
}
extern "C" void
dino_plugins_rtp_voice_processor_analyze_reverse_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
int err;
#if defined(WEBRTC0)
webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
webrtc::AudioProcessing *apm = native->apm;
GstMapInfo map;
gst_buffer_map(buffer, &map, GST_MAP_READ);
webrtc::AudioFrame frame;
frame.num_channels_ = info->channels;
frame.sample_rate_hz_ = info->rate;
frame.samples_per_channel_ = gst_buffer_get_size(buffer) / info->bpf;
memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf);
apm->set_stream_delay_ms (native->stream_delay);
err = apm->AnalyzeReverseStream(&frame);
gst_buffer_unmap(buffer, &map);
#elif defined(WEBRTC1) || defined(WEBRTC2)
rtc::scoped_refptr<webrtc::AudioProcessing> apm = native->apm;
#ifdef WEBRTC1
webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
#else
webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS);
#endif
GstAudioBuffer abuf;
if (!gst_audio_buffer_map (&abuf, info, buffer, GST_MAP_READWRITE)) {
g_warning("voice_processor_native.cpp: analyze_reverse_stream: gst_audio_buffer_map failed");
return;
}
apm->set_stream_delay_ms (native->stream_delay);
auto * const data = (int16_t * const) abuf.planes[0];
err = apm->ProcessReverseStream (data, config, config, data);
gst_audio_buffer_unmap (&abuf);
#endif
if (err < 0) g_warning("voice_processor_native.cpp: ProcessReverseStream %i", err);
}
extern "C" void dino_plugins_rtp_voice_processor_notify_gain_level(void *native_ptr, gint gain_level) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
#if defined(WEBRTC0)
webrtc::AudioProcessing *apm = native->apm;
apm->gain_control()->set_stream_analog_level(gain_level);
#elif defined(WEBRTC1) || defined(WEBRTC2)
rtc::scoped_refptr<webrtc::AudioProcessing> apm = native->apm;
apm->set_stream_analog_level(gain_level);
#endif
}
extern "C" gint dino_plugins_rtp_voice_processor_get_suggested_gain_level(void *native_ptr) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
int level = 0;
#if defined(WEBRTC0)
webrtc::AudioProcessing *apm = native->apm;
level = apm->gain_control()->stream_analog_level();
#elif defined(WEBRTC1) || defined(WEBRTC2)
rtc::scoped_refptr<webrtc::AudioProcessing> apm = native->apm;
level = apm->recommended_stream_analog_level();
#endif
return level;
}
extern "C" bool dino_plugins_rtp_voice_processor_get_stream_has_voice(void *native_ptr) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
bool has_voice = false;
#if defined(WEBRTC0)
webrtc::AudioProcessing *apm = native->apm;
has_voice = apm->voice_detection()->stream_has_voice();
#elif defined(WEBRTC1)
rtc::scoped_refptr<webrtc::AudioProcessing> apm = native->apm;
webrtc::AudioProcessingStats stats = apm->GetStatistics ();
has_voice = stats.voice_detected.value_or(false);
#endif
return has_voice;
}
extern "C" void dino_plugins_rtp_voice_processor_set_stream_delay(void *native_ptr, gint stream_delay) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
native->stream_delay = stream_delay;
}
extern "C" void dino_plugins_rtp_voice_processor_adjust_stream_delay(void *native_ptr) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
int median, std, poor_delays;
float fraction_poor_delays;
#if defined(WEBRTC0)
webrtc::AudioProcessing *apm = native->apm;
apm->echo_cancellation()->GetDelayMetrics(&median, &std, &fraction_poor_delays);
poor_delays = (int)(fraction_poor_delays * 100.0);
#elif defined(WEBRTC1) || defined(WEBRTC2)
rtc::scoped_refptr<webrtc::AudioProcessing> apm = native->apm;
webrtc::AudioProcessingStats stats = apm->GetStatistics();
median = stats.delay_median_ms.value_or(-1);
std = stats.delay_standard_deviation_ms.value_or(-1);
fraction_poor_delays = (float) stats.divergent_filter_fraction.value_or(-1.0);
poor_delays = (int) (fraction_poor_delays * 100.0);
#endif
if (fraction_poor_delays < 0 || (native->last_median == median && native->last_poor_delays == poor_delays)) return;
g_debug("voice_processor_native.cpp: Stream delay metrics: median=%i std=%i poor_delays=%i%%", median, std, poor_delays);
native->last_median = median;
native->last_poor_delays = poor_delays;
#if defined(WEBRTC0)
if (poor_delays > 90) {
native->stream_delay = std::min(std::max(0, native->stream_delay + std::min(48, std::max(median, -48))), 384);
g_debug("voice_processor_native.cpp: set stream_delay=%i", native->stream_delay);
}
#endif
}
extern "C" void
dino_plugins_rtp_voice_processor_process_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
int err;
#if defined(WEBRTC0)
webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
webrtc::AudioProcessing *apm = native->apm;
GstMapInfo map;
gst_buffer_map(buffer, &map, GST_MAP_READWRITE);
webrtc::AudioFrame frame;
frame.num_channels_ = info->channels;
frame.sample_rate_hz_ = info->rate;
frame.samples_per_channel_ = info->rate / 100;
memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf);
err = apm->ProcessStream(&frame);
if (err >= 0) memcpy(map.data, frame.data_, frame.samples_per_channel_ * info->bpf);
gst_buffer_unmap(buffer, &map);
#elif defined(WEBRTC1) || defined(WEBRTC2)
rtc::scoped_refptr<webrtc::AudioProcessing> apm = native->apm;
#ifdef WEBRTC1
webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
#else
webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS);
#endif
GstAudioBuffer abuf;
if (!gst_audio_buffer_map (&abuf, info, buffer, GST_MAP_READWRITE)) {
g_warning("voice_processor_native.cpp: process_stream: gst_audio_buffer_map failed");
return;
}
auto * const data = (int16_t * const) abuf.planes[0];
err = apm->ProcessStream (data, config, config, data);
gst_audio_buffer_unmap (&abuf);
#endif
if (err < 0) g_warning("voice_processor_native.cpp: ProcessStream %i", err);
}
extern "C" void dino_plugins_rtp_voice_processor_destroy_native(void *native_ptr) {
auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
#if defined(WEBRTC0)
delete native->apm;
native->apm = NULL;
#elif defined(WEBRTC1) || defined(WEBRTC2)
native->apm = nullptr;
#endif
delete native;
}
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