Move OMEMO, OpenPGP and RTP plugins into main/libdino

1 files changed, 256 insertions, 0 deletions

diff --git a/main/src/plugins/rtp/voice_processor_native.cpp b/main/src/plugins/rtp/voice_processor_native.cpp
new file mode 100644
index 00000000..71cc8a54
--- /dev/null
+++ b/main/src/plugins/rtp/voice_processor_native.cpp
@@ -0,0 +1,256 @@
+#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();
+    native->stream_delay = stream_delay;
+#if defined(WEBRTC0)
+    webrtc::Config config;
+    config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
+    config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, 85));
+    native->apm = webrtc::AudioProcessing::Create(config);
+
+    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.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
+    return native;
+}
+
+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);
+
+    apm->set_stream_delay_ms(native->stream_delay);
+    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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