diff options
Diffstat (limited to 'plugins/rtp/src/voice_processor_native.cpp')
| -rw-r--r-- | plugins/rtp/src/voice_processor_native.cpp | 141 |
1 files changed, 141 insertions, 0 deletions
diff --git a/plugins/rtp/src/voice_processor_native.cpp b/plugins/rtp/src/voice_processor_native.cpp new file mode 100644 index 00000000..9b3292b8 --- /dev/null +++ b/plugins/rtp/src/voice_processor_native.cpp @@ -0,0 +1,141 @@ +#include <algorithm> +#include <gst/gst.h> +#include <gst/audio/audio.h> +#include <webrtc/modules/audio_processing/include/audio_processing.h> +#include <webrtc/modules/interface/module_common_types.h> +#include <webrtc/system_wrappers/include/trace.h> + +#define SAMPLE_RATE 48000 +#define SAMPLE_CHANNELS 1 + +struct _DinoPluginsRtpVoiceProcessorNative { + webrtc::AudioProcessing *apm; + gint stream_delay; +}; + +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) { + _DinoPluginsRtpVoiceProcessorNative *native = new _DinoPluginsRtpVoiceProcessorNative(); + 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); + native->stream_delay = stream_delay; + return native; +} + +extern "C" void dino_plugins_rtp_voice_processor_setup_native(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + 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); +} + +extern "C" void +dino_plugins_rtp_voice_processor_analyze_reverse_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false); + webrtc::AudioProcessing *apm = native->apm; + + GstAudioBuffer audio_buffer; + gst_audio_buffer_map(&audio_buffer, info, buffer, 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_, audio_buffer.planes[0], frame.samples_per_channel_ * info->bpf); + + int err = apm->AnalyzeReverseStream(&frame); + if (err < 0) g_warning("ProcessReverseStream %i", err); + + gst_audio_buffer_unmap(&audio_buffer); +} + +extern "C" void dino_plugins_rtp_voice_processor_notify_gain_level(void *native_ptr, gint gain_level) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + apm->gain_control()->set_stream_analog_level(gain_level); +} + +extern "C" gint dino_plugins_rtp_voice_processor_get_suggested_gain_level(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + return apm->gain_control()->stream_analog_level(); +} + +extern "C" bool dino_plugins_rtp_voice_processor_get_stream_has_voice(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + return apm->voice_detection()->stream_has_voice(); +} + +extern "C" void dino_plugins_rtp_voice_processor_adjust_stream_delay(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::AudioProcessing *apm = native->apm; + int median, std; + float fraction_poor_delays; + apm->echo_cancellation()->GetDelayMetrics(&median, &std, &fraction_poor_delays); + if (fraction_poor_delays < 0) return; + g_debug("voice_processor_native.cpp: Stream delay metrics: %i %i %f", median, std, fraction_poor_delays); + if (fraction_poor_delays > 0.5) { + native->stream_delay = std::max(0, native->stream_delay + std::min(-10, std::max(median, 10))); + g_debug("Adjusted stream delay %i", native->stream_delay); + } +} + +extern "C" void +dino_plugins_rtp_voice_processor_process_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false); + webrtc::AudioProcessing *apm = native->apm; + + GstAudioBuffer audio_buffer; + gst_audio_buffer_map(&audio_buffer, info, buffer, 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_, audio_buffer.planes[0], frame.samples_per_channel_ * info->bpf); + + apm->set_stream_delay_ms(native->stream_delay); + int err = apm->ProcessStream(&frame); + if (err >= 0) memcpy(audio_buffer.planes[0], frame.data_, frame.samples_per_channel_ * info->bpf); + if (err < 0) g_warning("ProcessStream %i", err); + + gst_audio_buffer_unmap(&audio_buffer); +} + +extern "C" void dino_plugins_rtp_voice_processor_destroy_native(void *native_ptr) { + _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr; + delete native; +}
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