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path: root/plugins/rtp/src/voice_processor_native.cpp
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Diffstat (limited to 'plugins/rtp/src/voice_processor_native.cpp')
-rw-r--r--plugins/rtp/src/voice_processor_native.cpp162
1 files changed, 138 insertions, 24 deletions
diff --git a/plugins/rtp/src/voice_processor_native.cpp b/plugins/rtp/src/voice_processor_native.cpp
index 8a052cf8..e41feb49 100644
--- a/plugins/rtp/src/voice_processor_native.cpp
+++ b/plugins/rtp/src/voice_processor_native.cpp
@@ -1,18 +1,28 @@
#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;
- gint stream_delay;
- gint last_median;
- gint last_poor_delays;
+#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) {
@@ -22,19 +32,20 @@ extern "C" void *dino_plugins_rtp_adjust_to_running_time(GstBaseTransform *trans
}
extern "C" void *dino_plugins_rtp_voice_processor_init_native(gint stream_delay) {
- _DinoPluginsRtpVoiceProcessorNative *native = new _DinoPluginsRtpVoiceProcessorNative();
+ 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;
- native->last_median = 0;
- native->last_poor_delays = 0;
return native;
}
extern "C" void dino_plugins_rtp_voice_processor_setup_native(void *native_ptr) {
- _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+ auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+#if defined(WEBRTC0)
webrtc::AudioProcessing *apm = native->apm;
webrtc::ProcessingConfig pconfig;
pconfig.streams[webrtc::ProcessingConfig::kInputStream] =
@@ -61,14 +72,36 @@ extern "C" void dino_plugins_rtp_voice_processor_setup_native(void *native_ptr)
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) {
- _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+ 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);
@@ -78,53 +111,111 @@ dino_plugins_rtp_voice_processor_analyze_reverse_stream(void *native_ptr, GstAud
frame.samples_per_channel_ = gst_buffer_get_size(buffer) / info->bpf;
memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf);
- int err = apm->AnalyzeReverseStream(&frame);
- if (err < 0) g_warning("voice_processor_native.cpp: ProcessReverseStream %i", err);
+ 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) {
- _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+ 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) {
- _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+ auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+ int level = 0;
+#if defined(WEBRTC0)
webrtc::AudioProcessing *apm = native->apm;
- return apm->gain_control()->stream_analog_level();
+ 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) {
- _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+ auto *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+ bool has_voice = false;
+#if defined(WEBRTC0)
webrtc::AudioProcessing *apm = native->apm;
- return apm->voice_detection()->stream_has_voice();
+ 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) {
- _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
- webrtc::AudioProcessing *apm = native->apm;
+ 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) {
- _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+ 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);
@@ -134,15 +225,38 @@ dino_plugins_rtp_voice_processor_process_stream(void *native_ptr, GstAudioInfo *
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);
- int err = apm->ProcessStream(&frame);
+ err = apm->ProcessStream(&frame);
if (err >= 0) memcpy(map.data, frame.data_, frame.samples_per_channel_ * info->bpf);
- if (err < 0) g_warning("voice_processor_native.cpp: ProcessStream %i", err);
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) {
- _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) 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;
} \ No newline at end of file
generated by cgit v1.3 (git 2.53.0) at 2026-11-04 13:04:10 +0000