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1.24.6

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This commit is contained in:
Wim Taymans 2024-08-21 12:52:07 +02:00
parent 851eb29c89
commit 3312612a46
8 changed files with 86 additions and 1245 deletions
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1
.gitignore vendored
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@ -91,3 +91,4 @@
/gst-plugins-bad-free-1.22.8.tar.xz
/gst-plugins-bad-free-1.22.9.tar.xz
/gst-plugins-bad-free-1.22.12.tar.xz
/gst-plugins-bad-free-1.24.6.tar.xz

881
0001-webrtcdsp-Update-code-for-webrtc-audio-processing-1.patch
View File

@ -1,881 +0,0 @@
From 0757dbc0b406ebd6c38c8d86bf4d4738a232db40 Mon Sep 17 00:00:00 2001
From: Arun Raghavan <arun@asymptotic.io>
Date: Wed, 2 Dec 2020 18:31:44 -0500
Subject: [PATCH 1/3] webrtcdsp: Update code for webrtc-audio-processing-1
Updated API usage appropriately, and now we have a versioned package to
track breaking vs. non-breaking updates.
Deprecates a number of properties (and we have to plug in our own values
for related enums which are now gone):
* echo-suprression-level
* experimental-agc
* extended-filter
* delay-agnostic
* voice-detection-frame-size-ms
* voice-detection-likelihood
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2943>
---
.../ext/webrtcdsp/gstwebrtcdsp.cpp | 271 +++++++-----------
.../ext/webrtcdsp/gstwebrtcechoprobe.cpp | 87 +++---
.../ext/webrtcdsp/gstwebrtcechoprobe.h | 9 +-
.../gst-plugins-bad/ext/webrtcdsp/meson.build | 4 +-
4 files changed, 164 insertions(+), 207 deletions(-)
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
index 7ee09488fb..c9a7cdae2f 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
@@ -71,9 +71,7 @@
#include "gstwebrtcdsp.h"
#include "gstwebrtcechoprobe.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>
+#include <modules/audio_processing/include/audio_processing.h>
GST_DEBUG_CATEGORY (webrtc_dsp_debug);
#define GST_CAT_DEFAULT (webrtc_dsp_debug)
@@ -82,10 +80,9 @@ GST_DEBUG_CATEGORY (webrtc_dsp_debug);
#define DEFAULT_COMPRESSION_GAIN_DB 9
#define DEFAULT_STARTUP_MIN_VOLUME 12
#define DEFAULT_LIMITER TRUE
-#define DEFAULT_GAIN_CONTROL_MODE webrtc::GainControl::kAdaptiveDigital
+#define DEFAULT_GAIN_CONTROL_MODE webrtc::AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital
#define DEFAULT_VOICE_DETECTION FALSE
#define DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS 10
-#define DEFAULT_VOICE_DETECTION_LIKELIHOOD webrtc::VoiceDetection::kLowLikelihood
static GstStaticPadTemplate gst_webrtc_dsp_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
@@ -119,7 +116,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
"channels = (int) [1, MAX]")
);
-typedef webrtc::EchoCancellation::SuppressionLevel GstWebrtcEchoSuppressionLevel;
+typedef int GstWebrtcEchoSuppressionLevel;
#define GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL \
(gst_webrtc_echo_suppression_level_get_type ())
static GType
@@ -127,10 +124,9 @@ gst_webrtc_echo_suppression_level_get_type (void)
{
static GType suppression_level_type = 0;
static const GEnumValue level_types[] = {
- {webrtc::EchoCancellation::kLowSuppression, "Low Suppression", "low"},
- {webrtc::EchoCancellation::kModerateSuppression,
- "Moderate Suppression", "moderate"},
- {webrtc::EchoCancellation::kHighSuppression, "high Suppression", "high"},
+ {1, "Low Suppression", "low"},
+ {2, "Moderate Suppression", "moderate"},
+ {3, "high Suppression", "high"},
{0, NULL, NULL}
};
@@ -141,7 +137,7 @@ gst_webrtc_echo_suppression_level_get_type (void)
return suppression_level_type;
}
-typedef webrtc::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
+typedef webrtc::AudioProcessing::Config::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
#define GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL \
(gst_webrtc_noise_suppression_level_get_type ())
static GType
@@ -149,10 +145,10 @@ gst_webrtc_noise_suppression_level_get_type (void)
{
static GType suppression_level_type = 0;
static const GEnumValue level_types[] = {
- {webrtc::NoiseSuppression::kLow, "Low Suppression", "low"},
- {webrtc::NoiseSuppression::kModerate, "Moderate Suppression", "moderate"},
- {webrtc::NoiseSuppression::kHigh, "High Suppression", "high"},
- {webrtc::NoiseSuppression::kVeryHigh, "Very High Suppression",
+ {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kLow, "Low Suppression", "low"},
+ {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kModerate, "Moderate Suppression", "moderate"},
+ {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh, "High Suppression", "high"},
+ {webrtc::AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh, "Very High Suppression",
"very-high"},
{0, NULL, NULL}
};
@@ -164,7 +160,7 @@ gst_webrtc_noise_suppression_level_get_type (void)
return suppression_level_type;
}
-typedef webrtc::GainControl::Mode GstWebrtcGainControlMode;
+typedef webrtc::AudioProcessing::Config::GainController1::Mode GstWebrtcGainControlMode;
#define GST_TYPE_WEBRTC_GAIN_CONTROL_MODE \
(gst_webrtc_gain_control_mode_get_type ())
static GType
@@ -172,8 +168,9 @@ gst_webrtc_gain_control_mode_get_type (void)
{
static GType gain_control_mode_type = 0;
static const GEnumValue mode_types[] = {
- {webrtc::GainControl::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
- {webrtc::GainControl::kFixedDigital, "Fixed Digital", "fixed-digital"},
+ {webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
+ {webrtc::AudioProcessing::Config::GainController1::kFixedDigital, "Fixed Digital", "fixed-digital"},
+ {webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog, "Adaptive Analog", "adaptive-analog"},
{0, NULL, NULL}
};
@@ -184,7 +181,7 @@ gst_webrtc_gain_control_mode_get_type (void)
return gain_control_mode_type;
}
-typedef webrtc::VoiceDetection::Likelihood GstWebrtcVoiceDetectionLikelihood;
+typedef int GstWebrtcVoiceDetectionLikelihood;
#define GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD \
(gst_webrtc_voice_detection_likelihood_get_type ())
static GType
@@ -192,10 +189,10 @@ gst_webrtc_voice_detection_likelihood_get_type (void)
{
static GType likelihood_type = 0;
static const GEnumValue likelihood_types[] = {
- {webrtc::VoiceDetection::kVeryLowLikelihood, "Very Low Likelihood", "very-low"},
- {webrtc::VoiceDetection::kLowLikelihood, "Low Likelihood", "low"},
- {webrtc::VoiceDetection::kModerateLikelihood, "Moderate Likelihood", "moderate"},
- {webrtc::VoiceDetection::kHighLikelihood, "High Likelihood", "high"},
+ {1, "Very Low Likelihood", "very-low"},
+ {2, "Low Likelihood", "low"},
+ {3, "Moderate Likelihood", "moderate"},
+ {4, "High Likelihood", "high"},
{0, NULL, NULL}
};
@@ -227,6 +224,7 @@ enum
PROP_VOICE_DETECTION,
PROP_VOICE_DETECTION_FRAME_SIZE_MS,
PROP_VOICE_DETECTION_LIKELIHOOD,
+ PROP_EXTRA_DELAY_MS,
};
/**
@@ -248,7 +246,7 @@ struct _GstWebrtcDsp
/* Protected by the stream lock */
GstAdapter *adapter;
GstPlanarAudioAdapter *padapter;
- webrtc::AudioProcessing * apm;
+ webrtc::AudioProcessing *apm;
/* Protected by the object lock */
gchar *probe_name;
@@ -257,21 +255,15 @@ struct _GstWebrtcDsp
/* Properties */
gboolean high_pass_filter;
gboolean echo_cancel;
- webrtc::EchoCancellation::SuppressionLevel echo_suppression_level;
gboolean noise_suppression;
- webrtc::NoiseSuppression::Level noise_suppression_level;
+ webrtc::AudioProcessing::Config::NoiseSuppression::Level noise_suppression_level;
gboolean gain_control;
- gboolean experimental_agc;
- gboolean extended_filter;
- gboolean delay_agnostic;
gint target_level_dbfs;
gint compression_gain_db;
gint startup_min_volume;
gboolean limiter;
- webrtc::GainControl::Mode gain_control_mode;
+ webrtc::AudioProcessing::Config::GainController1::Mode gain_control_mode;
gboolean voice_detection;
- gint voice_detection_frame_size_ms;
- webrtc::VoiceDetection::Likelihood voice_detection_likelihood;
};
G_DEFINE_TYPE_WITH_CODE (GstWebrtcDsp, gst_webrtc_dsp, GST_TYPE_AUDIO_FILTER,
@@ -376,9 +368,9 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
GstClockTime rec_time)
{
GstWebrtcEchoProbe *probe = NULL;
- webrtc::AudioProcessing * apm;
- webrtc::AudioFrame frame;
+ webrtc::AudioProcessing *apm;
GstBuffer *buf = NULL;
+ GstAudioBuffer abuf;
GstFlowReturn ret = GST_FLOW_OK;
gint err, delay;
@@ -391,48 +383,44 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
if (!probe)
return GST_FLOW_OK;
+ webrtc::StreamConfig config (probe->info.rate, probe->info.channels,
+ false);
apm = self->apm;
- if (self->delay_agnostic)
- rec_time = GST_CLOCK_TIME_NONE;
-
-again:
- delay = gst_webrtc_echo_probe_read (probe, rec_time, (gpointer) &frame, &buf);
+ delay = gst_webrtc_echo_probe_read (probe, rec_time, &buf);
apm->set_stream_delay_ms (delay);
+ g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
+
if (delay < 0)
goto done;
- if (frame.sample_rate_hz_ != self->info.rate) {
+ if (probe->info.rate != self->info.rate) {
GST_ELEMENT_ERROR (self, STREAM, FORMAT,
("Echo Probe has rate %i , while the DSP is running at rate %i,"
" use a caps filter to ensure those are the same.",
- frame.sample_rate_hz_, self->info.rate), (NULL));
+ probe->info.rate, self->info.rate), (NULL));
ret = GST_FLOW_ERROR;
goto done;
}
- if (buf) {
- webrtc::StreamConfig config (frame.sample_rate_hz_, frame.num_channels_,
- false);
- GstAudioBuffer abuf;
- float * const * data;
+ gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
+
+ if (probe->interleaved) {
+ int16_t * const data = (int16_t * const) abuf.planes[0];
- gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
- data = (float * const *) abuf.planes;
if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
webrtc_error_to_string (err));
- gst_audio_buffer_unmap (&abuf);
- gst_buffer_replace (&buf, NULL);
} else {
- if ((err = apm->AnalyzeReverseStream (&frame)) < 0)
+ float * const * data = (float * const *) abuf.planes;
+
+ if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
webrtc_error_to_string (err));
}
- if (self->delay_agnostic)
- goto again;
+ gst_audio_buffer_unmap (&abuf);
done:
gst_object_unref (probe);
@@ -443,16 +431,14 @@ done:
static void
gst_webrtc_vad_post_activity (GstWebrtcDsp *self, GstBuffer *buffer,
- gboolean stream_has_voice)
+ gboolean stream_has_voice, guint8 level)
{
GstClockTime timestamp = GST_BUFFER_PTS (buffer);
GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (self);
GstStructure *s;
GstClockTime stream_time;
GstAudioLevelMeta *meta;
- guint8 level;
- level = self->apm->level_estimator ()->RMS ();
meta = gst_buffer_get_audio_level_meta (buffer);
if (meta) {
meta->voice_activity = stream_has_voice;
@@ -481,6 +467,7 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
{
GstAudioBuffer abuf;
webrtc::AudioProcessing * apm = self->apm;
+ webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
gint err;
if (!gst_audio_buffer_map (&abuf, &self->info, buffer,
@@ -490,19 +477,10 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
}
if (self->interleaved) {
- webrtc::AudioFrame frame;
- frame.num_channels_ = self->info.channels;
- frame.sample_rate_hz_ = self->info.rate;
- frame.samples_per_channel_ = self->period_samples;
-
- memcpy (frame.data_, abuf.planes[0], self->period_size);
- err = apm->ProcessStream (&frame);
- if (err >= 0)
- memcpy (abuf.planes[0], frame.data_, self->period_size);
+ int16_t * const data = (int16_t * const) abuf.planes[0];
+ err = apm->ProcessStream (data, config, config, data);
} else {
float * const * data = (float * const *) abuf.planes;
- webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
-
err = apm->ProcessStream (data, config, config, data);
}
@@ -511,10 +489,13 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
webrtc_error_to_string (err));
} else {
if (self->voice_detection) {
- gboolean stream_has_voice = apm->voice_detection ()->stream_has_voice ();
+ webrtc::AudioProcessingStats stats = apm->GetStatistics ();
+ gboolean stream_has_voice = stats.voice_detected && *stats.voice_detected;
+ // The meta takes the value as -dbov, so we negate
+ guint8 level = stats.output_rms_dbfs ? (guint8) -(*stats.output_rms_dbfs) : 127;
if (stream_has_voice != self->stream_has_voice)
- gst_webrtc_vad_post_activity (self, buffer, stream_has_voice);
+ gst_webrtc_vad_post_activity (self, buffer, stream_has_voice, level);
self->stream_has_voice = stream_has_voice;
}
@@ -583,21 +564,9 @@ static gboolean
gst_webrtc_dsp_start (GstBaseTransform * btrans)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
- webrtc::Config config;
GST_OBJECT_LOCK (self);
- config.Set < webrtc::ExtendedFilter >
- (new webrtc::ExtendedFilter (self->extended_filter));
- config.Set < webrtc::ExperimentalAgc >
- (new webrtc::ExperimentalAgc (self->experimental_agc, self->startup_min_volume));
- config.Set < webrtc::DelayAgnostic >
- (new webrtc::DelayAgnostic (self->delay_agnostic));
-
- /* TODO Intelligibility enhancer, Beamforming, etc. */
-
- self->apm = webrtc::AudioProcessing::Create (config);
-
if (self->echo_cancel) {
self->probe = gst_webrtc_acquire_echo_probe (self->probe_name);
@@ -618,10 +587,8 @@ static gboolean
gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
{
GstWebrtcDsp *self = GST_WEBRTC_DSP (filter);
- webrtc::AudioProcessing * apm;
- webrtc::ProcessingConfig pconfig;
+ webrtc::AudioProcessing::Config config;
GstAudioInfo probe_info = *info;
- gint err = 0;
GST_LOG_OBJECT (self, "setting format to %s with %i Hz and %i channels",
info->finfo->description, info->rate, info->channels);
@@ -633,7 +600,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
self->info = *info;
self->interleaved = (info->layout == GST_AUDIO_LAYOUT_INTERLEAVED);
- apm = self->apm;
+ self->apm = webrtc::AudioProcessingBuilder().Create();
if (!self->interleaved)
gst_planar_audio_adapter_configure (self->padapter, info);
@@ -642,8 +609,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
self->period_samples = info->rate / 100;
self->period_size = self->period_samples * info->bpf;
- if (self->interleaved &&
- (webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
+ if (self->interleaved && (self->period_size > MAX_DATA_SIZE_SAMPLES * 2))
goto period_too_big;
if (self->probe) {
@@ -658,40 +624,31 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
GST_WEBRTC_ECHO_PROBE_UNLOCK (self->probe);
}
- /* input stream */
- pconfig.streams[webrtc::ProcessingConfig::kInputStream] =
- webrtc::StreamConfig (info->rate, info->channels, false);
- /* output stream */
- pconfig.streams[webrtc::ProcessingConfig::kOutputStream] =
- webrtc::StreamConfig (info->rate, info->channels, false);
- /* reverse input stream */
- pconfig.streams[webrtc::ProcessingConfig::kReverseInputStream] =
- webrtc::StreamConfig (probe_info.rate, probe_info.channels, false);
- /* reverse output stream */
- pconfig.streams[webrtc::ProcessingConfig::kReverseOutputStream] =
- webrtc::StreamConfig (probe_info.rate, probe_info.channels, false);
-
- if ((err = apm->Initialize (pconfig)) < 0)
- goto initialize_failed;
-
/* Setup Filters */
+ // TODO: expose pre_amplifier
+
if (self->high_pass_filter) {
GST_DEBUG_OBJECT (self, "Enabling High Pass filter");
- apm->high_pass_filter ()->Enable (true);
+ config.high_pass_filter.enabled = true;
}
if (self->echo_cancel) {
GST_DEBUG_OBJECT (self, "Enabling Echo Cancellation");
- apm->echo_cancellation ()->enable_drift_compensation (false);
- apm->echo_cancellation ()
- ->set_suppression_level (self->echo_suppression_level);
- apm->echo_cancellation ()->Enable (true);
+ config.echo_canceller.enabled = true;
}
if (self->noise_suppression) {
GST_DEBUG_OBJECT (self, "Enabling Noise Suppression");
- apm->noise_suppression ()->set_level (self->noise_suppression_level);
- apm->noise_suppression ()->Enable (true);
+ config.noise_suppression.enabled = true;
+ config.noise_suppression.level = self->noise_suppression_level;
+ }
+
+ // TODO: expose transient suppression
+
+ if (self->voice_detection) {
+ GST_DEBUG_OBJECT (self, "Enabling Voice Activity Detection");
+ config.voice_detection.enabled = true;
+ self->stream_has_voice = FALSE;
}
if (self->gain_control) {
@@ -706,30 +663,17 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
g_type_class_unref (mode_class);
- apm->gain_control ()->set_mode (self->gain_control_mode);
- apm->gain_control ()->set_target_level_dbfs (self->target_level_dbfs);
- apm->gain_control ()->set_compression_gain_db (self->compression_gain_db);
- apm->gain_control ()->enable_limiter (self->limiter);
- apm->gain_control ()->Enable (true);
+ config.gain_controller1.enabled = true;
+ config.gain_controller1.target_level_dbfs = self->target_level_dbfs;
+ config.gain_controller1.compression_gain_db = self->compression_gain_db;
+ config.gain_controller1.enable_limiter = self->limiter;
+ config.level_estimation.enabled = true;
}
- if (self->voice_detection) {
- GEnumClass *likelihood_class = (GEnumClass *)
- g_type_class_ref (GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD);
- GST_DEBUG_OBJECT (self, "Enabling Voice Activity Detection, frame size "
- "%d milliseconds, likelihood: %s", self->voice_detection_frame_size_ms,
- g_enum_get_value (likelihood_class,
- self->voice_detection_likelihood)->value_name);
- g_type_class_unref (likelihood_class);
+ // TODO: expose gain controller 2
+ // TODO: expose residual echo detector
- self->stream_has_voice = FALSE;
-
- apm->voice_detection ()->Enable (true);
- apm->voice_detection ()->set_likelihood (self->voice_detection_likelihood);
- apm->voice_detection ()->set_frame_size_ms (
- self->voice_detection_frame_size_ms);
- apm->level_estimator ()->Enable (true);
- }
+ self->apm->ApplyConfig (config);
GST_OBJECT_UNLOCK (self);
@@ -738,9 +682,9 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
period_too_big:
GST_OBJECT_UNLOCK (self);
GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period "
- "(maximum is %" G_GSIZE_FORMAT " samples and we have %u samples), "
+ "(maximum is %d samples and we have %u samples), "
"reduce the number of channels or the rate.",
- webrtc::AudioFrame::kMaxDataSizeSamples, self->period_size / 2);
+ MAX_DATA_SIZE_SAMPLES, self->period_size / 2);
return FALSE;
probe_has_wrong_rate:
@@ -751,14 +695,6 @@ probe_has_wrong_rate:
" use a caps filter to ensure those are the same.",
probe_info.rate, info->rate), (NULL));
return FALSE;
-
-initialize_failed:
- GST_OBJECT_UNLOCK (self);
- GST_ELEMENT_ERROR (self, LIBRARY, INIT,
- ("Failed to initialize WebRTC Audio Processing library"),
- ("webrtc::AudioProcessing::Initialize() failed: %s",
- webrtc_error_to_string (err)));
- return FALSE;
}
static gboolean
@@ -803,8 +739,6 @@ gst_webrtc_dsp_set_property (GObject * object,
self->echo_cancel = g_value_get_boolean (value);
break;
case PROP_ECHO_SUPPRESSION_LEVEL:
- self->echo_suppression_level =
- (GstWebrtcEchoSuppressionLevel) g_value_get_enum (value);
break;
case PROP_NOISE_SUPPRESSION:
self->noise_suppression = g_value_get_boolean (value);
@@ -817,13 +751,10 @@ gst_webrtc_dsp_set_property (GObject * object,
self->gain_control = g_value_get_boolean (value);
break;
case PROP_EXPERIMENTAL_AGC:
- self->experimental_agc = g_value_get_boolean (value);
break;
case PROP_EXTENDED_FILTER:
- self->extended_filter = g_value_get_boolean (value);
break;
case PROP_DELAY_AGNOSTIC:
- self->delay_agnostic = g_value_get_boolean (value);
break;
case PROP_TARGET_LEVEL_DBFS:
self->target_level_dbfs = g_value_get_int (value);
@@ -845,11 +776,8 @@ gst_webrtc_dsp_set_property (GObject * object,
self->voice_detection = g_value_get_boolean (value);
break;
case PROP_VOICE_DETECTION_FRAME_SIZE_MS:
- self->voice_detection_frame_size_ms = g_value_get_int (value);
break;
case PROP_VOICE_DETECTION_LIKELIHOOD:
- self->voice_detection_likelihood =
- (GstWebrtcVoiceDetectionLikelihood) g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -876,7 +804,7 @@ gst_webrtc_dsp_get_property (GObject * object,
g_value_set_boolean (value, self->echo_cancel);
break;
case PROP_ECHO_SUPPRESSION_LEVEL:
- g_value_set_enum (value, self->echo_suppression_level);
+ g_value_set_enum (value, (GstWebrtcEchoSuppressionLevel) 2);
break;
case PROP_NOISE_SUPPRESSION:
g_value_set_boolean (value, self->noise_suppression);
@@ -888,13 +816,13 @@ gst_webrtc_dsp_get_property (GObject * object,
g_value_set_boolean (value, self->gain_control);
break;
case PROP_EXPERIMENTAL_AGC:
- g_value_set_boolean (value, self->experimental_agc);
+ g_value_set_boolean (value, false);
break;
case PROP_EXTENDED_FILTER:
- g_value_set_boolean (value, self->extended_filter);
+ g_value_set_boolean (value, false);
break;
case PROP_DELAY_AGNOSTIC:
- g_value_set_boolean (value, self->delay_agnostic);
+ g_value_set_boolean (value, false);
break;
case PROP_TARGET_LEVEL_DBFS:
g_value_set_int (value, self->target_level_dbfs);
@@ -915,10 +843,10 @@ gst_webrtc_dsp_get_property (GObject * object,
g_value_set_boolean (value, self->voice_detection);
break;
case PROP_VOICE_DETECTION_FRAME_SIZE_MS:
- g_value_set_int (value, self->voice_detection_frame_size_ms);
+ g_value_set_int (value, 0);
break;
case PROP_VOICE_DETECTION_LIKELIHOOD:
- g_value_set_enum (value, self->voice_detection_likelihood);
+ g_value_set_enum (value, 2);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -1005,13 +933,13 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
g_object_class_install_property (gobject_class,
PROP_ECHO_SUPPRESSION_LEVEL,
- g_param_spec_enum ("echo-suppression-level", "Echo Suppression Level",
+ g_param_spec_enum ("echo-suppression-level",
+ "Echo Suppression Level (does nothing)",
"Controls the aggressiveness of the suppressor. A higher level "
"trades off double-talk performance for increased echo suppression.",
- GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL,
- webrtc::EchoCancellation::kModerateSuppression,
+ GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL, 2,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
- G_PARAM_CONSTRUCT)));
+ G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
g_object_class_install_property (gobject_class,
PROP_NOISE_SUPPRESSION,
@@ -1026,7 +954,7 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
"Controls the aggressiveness of the suppression. Increasing the "
"level will reduce the noise level at the expense of a higher "
"speech distortion.", GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL,
- webrtc::EchoCancellation::kModerateSuppression,
+ webrtc::AudioProcessing::Config::NoiseSuppression::Level::kModerate,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
@@ -1039,24 +967,26 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
g_object_class_install_property (gobject_class,
PROP_EXPERIMENTAL_AGC,
- g_param_spec_boolean ("experimental-agc", "Experimental AGC",
+ g_param_spec_boolean ("experimental-agc",
+ "Experimental AGC (does nothing)",
"Enable or disable experimental automatic gain control.",
FALSE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
- G_PARAM_CONSTRUCT)));
+ G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
g_object_class_install_property (gobject_class,
PROP_EXTENDED_FILTER,
g_param_spec_boolean ("extended-filter", "Extended Filter",
"Enable or disable the extended filter.",
TRUE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
- G_PARAM_CONSTRUCT)));
+ G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
g_object_class_install_property (gobject_class,
PROP_DELAY_AGNOSTIC,
- g_param_spec_boolean ("delay-agnostic", "Delay Agnostic",
+ g_param_spec_boolean ("delay-agnostic",
+ "Delay agnostic mode (does nothing)",
"Enable or disable the delay agnostic mode.",
FALSE, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
- G_PARAM_CONSTRUCT)));
+ G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
g_object_class_install_property (gobject_class,
PROP_TARGET_LEVEL_DBFS,
@@ -1111,24 +1041,23 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
g_object_class_install_property (gobject_class,
PROP_VOICE_DETECTION_FRAME_SIZE_MS,
g_param_spec_int ("voice-detection-frame-size-ms",
- "Voice Detection Frame Size Milliseconds",
+ "Voice detection frame size in milliseconds (does nothing)",
"Sets the |size| of the frames in ms on which the VAD will operate. "
"Larger frames will improve detection accuracy, but reduce the "
"frequency of updates",
10, 30, DEFAULT_VOICE_DETECTION_FRAME_SIZE_MS,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
- G_PARAM_CONSTRUCT)));
+ G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
g_object_class_install_property (gobject_class,
PROP_VOICE_DETECTION_LIKELIHOOD,
g_param_spec_enum ("voice-detection-likelihood",
- "Voice Detection Likelihood",
+ "Voice detection likelihood (does nothing)",
"Specifies the likelihood that a frame will be declared to contain "
"voice.",
- GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD,
- DEFAULT_VOICE_DETECTION_LIKELIHOOD,
+ GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD, 2,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
- G_PARAM_CONSTRUCT)));
+ G_PARAM_CONSTRUCT | G_PARAM_DEPRECATED)));
gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_GAIN_CONTROL_MODE, (GstPluginAPIFlags) 0);
gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL, (GstPluginAPIFlags) 0);
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
index acdb3d8a7d..8e8ca064c4 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
@@ -33,7 +33,8 @@
#include "gstwebrtcechoprobe.h"
-#include <webrtc/modules/interface/module_common_types.h>
+#include <modules/audio_processing/include/audio_processing.h>
+
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_EXTERN (webrtc_dsp_debug);
@@ -102,7 +103,7 @@ gst_webrtc_echo_probe_setup (GstAudioFilter * filter, const GstAudioInfo * info)
self->period_size = self->period_samples * info->bpf;
if (self->interleaved &&
- (webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
+ (MAX_DATA_SIZE_SAMPLES * 2) < self->period_size)
goto period_too_big;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
@@ -112,9 +113,9 @@ gst_webrtc_echo_probe_setup (GstAudioFilter * filter, const GstAudioInfo * info)
period_too_big:
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period "
- "(maximum is %" G_GSIZE_FORMAT " samples and we have %u samples), "
+ "(maximum is %d samples and we have %u samples), "
"reduce the number of channels or the rate.",
- webrtc::AudioFrame::kMaxDataSizeSamples, self->period_size / 2);
+ MAX_DATA_SIZE_SAMPLES, self->period_size / 2);
return FALSE;
}
@@ -303,18 +304,20 @@ gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe)
gint
gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
- gpointer _frame, GstBuffer ** buf)
+ GstBuffer ** buf)
{
- webrtc::AudioFrame * frame = (webrtc::AudioFrame *) _frame;
GstClockTimeDiff diff;
- gsize avail, skip, offset, size;
+ gsize avail, skip, offset, size = 0;
gint delay = -1;
GST_WEBRTC_ECHO_PROBE_LOCK (self);
+ /* We always return a buffer -- if don't have data (size == 0), we generate a
+ * silence buffer */
+
if (!GST_CLOCK_TIME_IS_VALID (self->latency) ||
!GST_AUDIO_INFO_IS_VALID (&self->info))
- goto done;
+ goto copy;
if (self->interleaved)
avail = gst_adapter_available (self->adapter) / self->info.bpf;
@@ -324,7 +327,7 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
/* In delay agnostic mode, just return 10ms of data */
if (!GST_CLOCK_TIME_IS_VALID (rec_time)) {
if (avail < self->period_samples)
- goto done;
+ goto copy;
size = self->period_samples;
skip = 0;
@@ -371,23 +374,51 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
size = MIN (avail - offset, self->period_samples - skip);
copy:
- if (self->interleaved) {
- skip *= self->info.bpf;
- offset *= self->info.bpf;
- size *= self->info.bpf;
-
- if (size < self->period_size)
- memset (frame->data_, 0, self->period_size);
-
- if (size) {
- gst_adapter_copy (self->adapter, (guint8 *) frame->data_ + skip,
- offset, size);
- gst_adapter_flush (self->adapter, offset + size);
- }
+ if (!size) {
+ /* No data, provide a period's worth of silence */
+ *buf = gst_buffer_new_allocate (NULL, self->period_size, NULL);
+ gst_buffer_memset (*buf, 0, 0, self->period_size);
+ gst_buffer_add_audio_meta (*buf, &self->info, self->period_samples,
+ NULL);
} else {
+ /* We have some actual data, pop period_samples' worth if have it, else pad
+ * with silence and provide what we do have */
GstBuffer *ret, *taken, *tmp;
- if (size) {
+ if (self->interleaved) {
+ skip *= self->info.bpf;
+ offset *= self->info.bpf;
+ size *= self->info.bpf;
+
+ gst_adapter_flush (self->adapter, offset);
+
+ /* we need to fill silence at the beginning and/or the end of the
+ * buffer in order to have period_samples in the buffer */
+ if (size < self->period_size) {
+ gsize padding = self->period_size - (skip + size);
+
+ taken = gst_adapter_take_buffer (self->adapter, size);
+ ret = gst_buffer_new ();
+
+ /* need some silence at the beginning */
+ if (skip) {
+ tmp = gst_buffer_new_allocate (NULL, skip, NULL);
+ gst_buffer_memset (tmp, 0, 0, skip);
+ ret = gst_buffer_append (ret, tmp);
+ }
+
+ ret = gst_buffer_append (ret, taken);
+
+ /* need some silence at the end */
+ if (padding) {
+ tmp = gst_buffer_new_allocate (NULL, padding, NULL);
+ gst_buffer_memset (tmp, 0, 0, padding);
+ ret = gst_buffer_append (ret, tmp);
+ }
+ } else {
+ ret = gst_adapter_take_buffer (self->adapter, size);
+ }
+ } else {
gst_planar_audio_adapter_flush (self->padapter, offset);
/* we need to fill silence at the beginning and/or the end of each
@@ -430,23 +461,13 @@ copy:
ret = gst_planar_audio_adapter_take_buffer (self->padapter, size,
GST_MAP_READWRITE);
}
- } else {
- ret = gst_buffer_new_allocate (NULL, self->period_size, NULL);
- gst_buffer_memset (ret, 0, 0, self->period_size);
- gst_buffer_add_audio_meta (ret, &self->info, self->period_samples,
- NULL);
}
*buf = ret;
}
- frame->num_channels_ = self->info.channels;
- frame->sample_rate_hz_ = self->info.rate;
- frame->samples_per_channel_ = self->period_samples;
-
delay = self->delay;
-done:
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return delay;
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
index 36fd34f179..488c0e958f 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
@@ -45,6 +45,12 @@ G_BEGIN_DECLS
#define GST_WEBRTC_ECHO_PROBE_LOCK(obj) g_mutex_lock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
#define GST_WEBRTC_ECHO_PROBE_UNLOCK(obj) g_mutex_unlock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
+/* From the webrtc audio_frame.h definition of kMaxDataSizeSamples:
+ * Stereo, 32 kHz, 120 ms (2 * 32 * 120)
+ * Stereo, 192 kHz, 20 ms (2 * 192 * 20)
+ */
+#define MAX_DATA_SIZE_SAMPLES 7680
+
typedef struct _GstWebrtcEchoProbe GstWebrtcEchoProbe;
typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass;
@@ -71,6 +77,7 @@ struct _GstWebrtcEchoProbe
GstClockTime latency;
gint delay;
gboolean interleaved;
+ gint extra_delay;
GstSegment segment;
GstAdapter *adapter;
@@ -92,7 +99,7 @@ GST_ELEMENT_REGISTER_DECLARE (webrtcechoprobe);
GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self,
- GstClockTime rec_time, gpointer frame, GstBuffer ** buf);
+ GstClockTime rec_time, GstBuffer ** buf);
G_END_DECLS
#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build b/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build
index 5aeae69a44..09565e27c7 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build
@@ -4,7 +4,7 @@ webrtc_sources = [
'gstwebrtcdspplugin.cpp'
]
-webrtc_dep = dependency('webrtc-audio-processing', version : ['>= 0.2', '< 0.4'],
+webrtc_dep = dependency('webrtc-audio-processing-1', version : ['>= 1.0'],
required : get_option('webrtcdsp'))
if not gnustl_dep.found() and get_option('webrtcdsp').enabled()
@@ -20,7 +20,7 @@ if webrtc_dep.found() and gnustl_dep.found()
dependencies : [gstbase_dep, gstaudio_dep, gstbadaudio_dep, webrtc_dep, gnustl_dep],
install : true,
install_dir : plugins_install_dir,
- override_options : ['cpp_std=c++11'],
+ override_options : ['cpp_std=c++17'],
)
plugins += [gstwebrtcdsp]
endif
--
2.45.2

36
0002-webrtcdsp-Map-probe-buffers-with-probe-info-not-dsp-.patch
View File

@ -1,36 +0,0 @@
From 187eb3d4874ff6ceccf85bed3e8161f68f8e09da Mon Sep 17 00:00:00 2001
From: Nirbheek Chauhan <nirbheek@centricular.com>
Date: Tue, 6 Jun 2023 16:34:19 +0530
Subject: [PATCH 2/3] webrtcdsp: Map probe buffers with probe info, not dsp
info
The probe's info may not precisely match the dsp's info. For instance,
the number of channels or their layout might be different.
```
GStreamer-Audio-CRITICAL **: 16:21:32.899: the GstAudioInfo argument is not equal to the GstAudioMeta's attached info
```
This broke in d5755744c3e2b70e9f04704ae9d18b928d9fa456.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
---
subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
index c9a7cdae2f..87e5cbc4a2 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
@@ -404,7 +404,7 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
goto done;
}
- gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
+ gst_audio_buffer_map (&abuf, &probe->info, buf, GST_MAP_READWRITE);
if (probe->interleaved) {
int16_t * const data = (int16_t * const) abuf.planes[0];
--
2.45.2

146
0003-webrtcdsp-Deal-with-echo-probe-info-not-being-availa.patch
View File

@ -1,146 +0,0 @@
From 7c5003367fe7528c5b5a5bac0243910e46019d0f Mon Sep 17 00:00:00 2001
From: Arun Raghavan <arun@asymptotic.io>
Date: Tue, 13 Jun 2023 11:41:43 -0400
Subject: [PATCH 3/3] webrtcdsp: Deal with echo probe info not being available
Even if we don't yet know what the echo probe format is, we want to be able to
provide silence for the reverse path, so that when the probe becomes available,
there is no ambiguity around what time period the new set of samples are for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
---
.../ext/webrtcdsp/gstwebrtcdsp.cpp | 20 ++++++++------
.../ext/webrtcdsp/gstwebrtcechoprobe.cpp | 26 +++++++++++++------
.../ext/webrtcdsp/gstwebrtcechoprobe.h | 3 ++-
3 files changed, 32 insertions(+), 17 deletions(-)
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
index 87e5cbc4a2..dc4190cff0 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.cpp
@@ -370,6 +370,8 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
GstWebrtcEchoProbe *probe = NULL;
webrtc::AudioProcessing *apm;
GstBuffer *buf = NULL;
+ GstAudioInfo info;
+ gboolean interleaved = self->interleaved;
GstAudioBuffer abuf;
GstFlowReturn ret = GST_FLOW_OK;
gint err, delay;
@@ -377,36 +379,38 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
GST_OBJECT_LOCK (self);
if (self->echo_cancel)
probe = GST_WEBRTC_ECHO_PROBE (g_object_ref (self->probe));
+ info = self->info;
GST_OBJECT_UNLOCK (self);
/* If echo cancellation is disabled */
if (!probe)
return GST_FLOW_OK;
- webrtc::StreamConfig config (probe->info.rate, probe->info.channels,
- false);
- apm = self->apm;
+ delay =
+ gst_webrtc_echo_probe_read (probe, rec_time, &buf, &info, &interleaved);
- delay = gst_webrtc_echo_probe_read (probe, rec_time, &buf);
+ apm = self->apm;
apm->set_stream_delay_ms (delay);
+ webrtc::StreamConfig config (info.rate, info.channels, false);
+
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
if (delay < 0)
goto done;
- if (probe->info.rate != self->info.rate) {
+ if (info.rate != self->info.rate) {
GST_ELEMENT_ERROR (self, STREAM, FORMAT,
("Echo Probe has rate %i , while the DSP is running at rate %i,"
" use a caps filter to ensure those are the same.",
- probe->info.rate, self->info.rate), (NULL));
+ info.rate, self->info.rate), (NULL));
ret = GST_FLOW_ERROR;
goto done;
}
- gst_audio_buffer_map (&abuf, &probe->info, buf, GST_MAP_READWRITE);
+ gst_audio_buffer_map (&abuf, &info, buf, GST_MAP_READWRITE);
- if (probe->interleaved) {
+ if (interleaved) {
int16_t * const data = (int16_t * const) abuf.planes[0];
if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
index 8e8ca064c4..13d0fc6cb0 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
@@ -304,7 +304,7 @@ gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe)
gint
gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
- GstBuffer ** buf)
+ GstBuffer ** buf, GstAudioInfo * info, gboolean * interleaved)
{
GstClockTimeDiff diff;
gsize avail, skip, offset, size = 0;
@@ -315,10 +315,17 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
/* We always return a buffer -- if don't have data (size == 0), we generate a
* silence buffer */
- if (!GST_CLOCK_TIME_IS_VALID (self->latency) ||
- !GST_AUDIO_INFO_IS_VALID (&self->info))
+ if (!GST_CLOCK_TIME_IS_VALID (self->latency))
goto copy;
+ /* If we have a format, use that, else generate silence in input format */
+ if (!GST_AUDIO_INFO_IS_VALID (&self->info)) {
+ goto copy;
+ } else {
+ *info = self->info;
+ *interleaved = self->interleaved;
+ }
+
if (self->interleaved)
avail = gst_adapter_available (self->adapter) / self->info.bpf;
else
@@ -375,11 +382,14 @@ gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
copy:
if (!size) {
- /* No data, provide a period's worth of silence */
- *buf = gst_buffer_new_allocate (NULL, self->period_size, NULL);
- gst_buffer_memset (*buf, 0, 0, self->period_size);
- gst_buffer_add_audio_meta (*buf, &self->info, self->period_samples,
- NULL);
+ /* No data, provide a period's worth of silence, using our format if we have
+ * it, or the provided format if we don't */
+ guint period_samples = info->rate / 100;
+ guint period_size = period_samples * info->bpf;
+
+ *buf = gst_buffer_new_allocate (NULL, period_size, NULL);
+ gst_buffer_memset (*buf, 0, 0, period_size);
+ gst_buffer_add_audio_meta (*buf, info, period_samples, NULL);
} else {
/* We have some actual data, pop period_samples' worth if have it, else pad
* with silence and provide what we do have */
diff --git a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
index 488c0e958f..30ab50b355 100644
--- a/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
+++ b/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.h
@@ -99,7 +99,8 @@ GST_ELEMENT_REGISTER_DECLARE (webrtcechoprobe);
GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self,
- GstClockTime rec_time, GstBuffer ** buf);
+ GstClockTime rec_time, GstBuffer ** buf, GstAudioInfo * info,
+ gboolean * interleaved);
G_END_DECLS
#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */
--
2.45.2

3
gst-p-bad-cleanup.sh
View File

@ -58,6 +58,7 @@ h264parse
hdvparse
hls
id3tag
insertbin
inter
interlace
invtelecine
@ -69,6 +70,7 @@ legacyresample
librfb
liveadder
midi
mse
mve
mpegdemux
mpeg4videoparse
@ -108,6 +110,7 @@ switchbin
timecode
transcode
tta
unixfd
valve
videofilters
videoframe_audiolevel

113
gstreamer1-plugins-bad-free.spec
View File

@ -5,14 +5,17 @@
%bcond extras %{defined fedora}
%bcond opencv %[ 0%{?fedora} >= 39 ]
%bcond openh264 %[ 0%{?fedora} >= 40 ]
%bcond svtav1 %[ 0%{?fedora} >= 40 ]
# requires new webrtc-audio-processing-1
%bcond webrtcdsp %[ 0%{?fedora} >= 40 || 0%{?rhel} >= 10 ]
#global gitrel 140
#global gitcommit 4ca3a22b6b33ad8be4383063e76f79c4d346535d
#global shortcommit %(c=%{gitcommit}; echo ${c:0:5})
Name: gstreamer1-plugins-bad-free
Version: 1.22.12
Release: 3%{?dist}
Version: 1.24.6
Release: 1%{?dist}
Summary: GStreamer streaming media framework "bad" plugins
License: LGPLv2+ and LGPLv2
@ -31,13 +34,6 @@ Source1: gst-p-bad-cleanup.sh
# https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5622
Patch: openh264-add-license-file.patch
# https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5780
Patch: openh264-drop-runtime-version-checks.patch
Patch: 0001-webrtcdsp-Update-code-for-webrtc-audio-processing-1.patch
Patch: 0002-webrtcdsp-Map-probe-buffers-with-probe-info-not-dsp-.patch
Patch: 0003-webrtcdsp-Deal-with-echo-probe-info-not-being-availa.patch
BuildRequires: meson >= 0.48.0
BuildRequires: gcc-c++
@ -83,11 +79,6 @@ BuildRequires: libwebp-devel
BuildRequires: mesa-libEGL-devel
BuildRequires: vulkan-devel
#BuildRequires: mesa-vulkan-devel
%if 0%{?fedora} >= 40 || 0%{?rhel} >= 10
BuildRequires: pkgconfig(webrtc-audio-processing-1)
%else
BuildRequires: pkgconfig(webrtc-audio-processing) >= 0.3
%endif
%if 0
BuildRequires: wpewebkit-devel
BuildRequires: wpebackend-fdo-devel
@ -95,6 +86,17 @@ BuildRequires: wpebackend-fdo-devel
BuildRequires: glslc
BuildRequires: libdrm-devel
BuildRequires: libva-devel
%if %{with svtav1}
BuildRequires: svt-av1-devel
%endif
BuildRequires: liblc3-devel
BuildRequires: json-glib-devel
%if %{with openh264}
BuildRequires: pkgconfig(openh264)
%endif
%if %{with webrtcdsp}
BuildRequires: pkgconfig(webrtc-audio-processing-1)
%endif
%if %{with extras}
BuildRequires: ladspa-devel
@ -117,7 +119,6 @@ BuildRequires: libcurl-devel
BuildRequires: libssh2-devel
BuildRequires: libxml2-devel
BuildRequires: game-music-emu-devel
BuildRequires: libkate-devel
BuildRequires: libmodplug-devel
BuildRequires: libmpcdec-devel
## Plugins not ported
@ -128,9 +129,6 @@ BuildRequires: openal-soft-devel
BuildRequires: opencv-devel
%endif
BuildRequires: openjpeg2-devel
%if %{with openh264}
BuildRequires: pkgconfig(openh264)
%endif
BuildRequires: pkgconfig(spandsp) >= 0.0.6
## Plugins not ported
#BuildRequires: SDL-devel
@ -144,7 +142,6 @@ BuildRequires: libnice-devel
BuildRequires: pkgconfig(ldacBT-enc)
%endif
BuildRequires: qrencode-devel
BuildRequires: json-glib-devel
BuildRequires: vo-amrwbenc-devel
BuildRequires: libavtp-devel
BuildRequires: libdca-devel
@ -310,11 +307,12 @@ aren't tested well enough, or the code is not of good enough quality.
%{!?with_extras:-D directsound=disabled -D dts=disabled } \
%{!?with_extras:-D fluidsynth=disabled -D openexr=disabled } \
%{!?with_extras:-D curl=disabled -D curl-ssh2=disabled } \
%{!?with_extras:-D ttml=disabled -D kate=disabled } \
%{!?with_extras:-D ttml=disabled } \
%{!?with_extras:-D modplug=disabled } \
%{!?with_extras:-D openal=disabled } \
%{!?with_opencv:-D opencv=disabled } \
%{!?with_openh264:-D openh264=disabled } \
%{!?with_svtav1:-D svtav1=disabled } \
%{!?with_extras:-D openjpeg=disabled } \
%{!?with_extras:-D wildmidi=disabled -D zbar=disabled } \
%{!?with_extras:-D gme=disabled -D lv2=disabled } \
@ -348,7 +346,9 @@ aren't tested well enough, or the code is not of good enough quality.
%{!?with_extras:-D qroverlay=disabled } \
-D gs=disabled -D isac=disabled \
-D onnx=disabled -D openaptx=disabled -Dgpl=enabled \
-D amfcodec=disabled -D directshow=disabled -D qsv=disabled
-D amfcodec=disabled -D directshow=disabled -D qsv=disabled \
%{!?with_webrtcdsp:-D webrtcdsp=disabled } \
-D aja=disabled -D qt6d3d11=disabled
%meson_build
@ -446,9 +446,6 @@ EOF
%find_lang gst-plugins-bad-%{majorminor}
# unpackaged files
rm $RPM_BUILD_ROOT%{_bindir}/playout
%ldconfig_scriptlets
%files -f gst-plugins-bad-%{majorminor}.lang
@ -459,9 +456,13 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_bindir}/gst-transcoder-%{majorminor}
# presets
%dir %{_datadir}/gstreamer-%{majorminor}/
%dir %{_datadir}/gstreamer-%{majorminor}/presets/
%{_datadir}/gstreamer-%{majorminor}/presets/GstFreeverb.prs
%dir %{_datadir}/gstreamer-%{majorminor}/encoding-profiles/
%dir %{_datadir}/gstreamer-%{majorminor}/encoding-profiles/device/
%{_datadir}/gstreamer-%{majorminor}/encoding-profiles/device/dvd.gep
%dir %{_datadir}/gstreamer-%{majorminor}/encoding-profiles/file-extension/
%{_datadir}/gstreamer-%{majorminor}/encoding-profiles/file-extension/avi.gep
%{_datadir}/gstreamer-%{majorminor}/encoding-profiles/file-extension/flv.gep
%{_datadir}/gstreamer-%{majorminor}/encoding-profiles/file-extension/mkv.gep
@ -471,6 +472,7 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_datadir}/gstreamer-%{majorminor}/encoding-profiles/file-extension/ogv.gep
%{_datadir}/gstreamer-%{majorminor}/encoding-profiles/file-extension/ts.gep
%{_datadir}/gstreamer-%{majorminor}/encoding-profiles/file-extension/webm.gep
%dir %{_datadir}/gstreamer-%{majorminor}/encoding-profiles/online-services/
%{_datadir}/gstreamer-%{majorminor}/encoding-profiles/online-services/youtube.gep
# Plugins without external dependencies
@ -478,6 +480,7 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_libdir}/gstreamer-%{majorminor}/libgstadpcmdec.so
%{_libdir}/gstreamer-%{majorminor}/libgstadpcmenc.so
%{_libdir}/gstreamer-%{majorminor}/libgstaiff.so
%{_libdir}/gstreamer-%{majorminor}/libgstanalyticsoverlay.so
%{_libdir}/gstreamer-%{majorminor}/libgstasfmux.so
%{_libdir}/gstreamer-%{majorminor}/libgstaudiobuffersplit.so
%{_libdir}/gstreamer-%{majorminor}/libgstaudiofxbad.so
@ -548,6 +551,11 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_libdir}/gstreamer-%{majorminor}/libgstvideosignal.so
%{_libdir}/gstreamer-%{majorminor}/libgstvmnc.so
%{_libdir}/gstreamer-%{majorminor}/libgsty4mdec.so
%{_libdir}/gstreamer-%{majorminor}/libgstcodec2json.so
%{_libdir}/gstreamer-%{majorminor}/libgstinsertbin.so
%{_libdir}/gstreamer-%{majorminor}/libgstmse.so
%{_libdir}/gstreamer-%{majorminor}/libgstunixfd.so
%{_libdir}/gstreamer-%{majorminor}/libgstuvcgadget.so
# System (Linux) specific plugins
%{_libdir}/gstreamer-%{majorminor}/libgstdvb.so
@ -568,6 +576,7 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_libdir}/gstreamer-%{majorminor}/libgstgsm.so
%{_libdir}/gstreamer-%{majorminor}/libgstgtkwayland.so
%{_libdir}/gstreamer-%{majorminor}/libgstkms.so
%{_libdir}/gstreamer-%{majorminor}/libgstlc3.so
%{_libdir}/gstreamer-%{majorminor}/libgstnvcodec.so
%{_libdir}/gstreamer-%{majorminor}/libgstopusparse.so
%{_libdir}/gstreamer-%{majorminor}/libgstrist.so
@ -581,7 +590,9 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_libdir}/gstreamer-%{majorminor}/libgstwaylandsink.so
%endif
%{_libdir}/gstreamer-%{majorminor}/libgstwebp.so
%if %{with webrtcdsp}
%{_libdir}/gstreamer-%{majorminor}/libgstwebrtcdsp.so
%endif
%if 0
%{_libdir}/gstreamer-%{majorminor}/libgstwpe.so
%endif
@ -590,6 +601,9 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_libdir}/gstreamer-%{majorminor}/libgstlv2.so
%{_libdir}/gstreamer-%{majorminor}/libgstttmlsubs.so
%endif
%if %{with svtav1}
%{_libdir}/gstreamer-%{majorminor}/libgstsvtav1.so
%endif
#debugging plugin
%{_libdir}/gstreamer-%{majorminor}/libgstdebugutilsbad.so
@ -611,7 +625,6 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_libdir}/gstreamer-%{majorminor}/libgstdtsdec.so
%{_libdir}/gstreamer-%{majorminor}/libgstflite.so
%{_libdir}/gstreamer-%{majorminor}/libgstgme.so
%{_libdir}/gstreamer-%{majorminor}/libgstkate.so
%{_libdir}/gstreamer-%{majorminor}/libgstladspa.so
%ifnarch s390x
%{_libdir}/gstreamer-%{majorminor}/libgstldac.so
@ -659,15 +672,18 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%files libs
%license COPYING
%{_libdir}/libgstanalytics-%{majorminor}.so.0{,.*}
%{_libdir}/libgstadaptivedemux-%{majorminor}.so.0{,.*}
%{_libdir}/libgstbasecamerabinsrc-%{majorminor}.so.0{,.*}
%{_libdir}/libgstbadaudio-%{majorminor}.so.0{,.*}
%{_libdir}/libgstcodecparsers-%{majorminor}.so.0{,.*}
%{_libdir}/libgstcodecs-%{majorminor}.so.0{,.*}
%{_libdir}/libgstcuda-%{majorminor}.so.0{,.*}
%{_libdir}/libgstdxva-%{majorminor}.so.0{,.*}
%{_libdir}/libgstinsertbin-%{majorminor}.so.0{,.*}
%{_libdir}/libgstisoff-%{majorminor}.so.0{,.*}
%{_libdir}/libgstmpegts-%{majorminor}.so.0{,.*}
%{_libdir}/libgstmse-%{majorminor}.so.0{,.*}
%{_libdir}/libgstplay-%{majorminor}.so.0{,.*}
%{_libdir}/libgstplayer-%{majorminor}.so.0{,.*}
%{_libdir}/libgstphotography-%{majorminor}.so.0{,.*}
@ -685,11 +701,14 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%endif
%{_libdir}/girepository-1.0/CudaGst-1.0.typelib
%{_libdir}/girepository-1.0/GstAnalytics-1.0.typelib
%{_libdir}/girepository-1.0/GstBadAudio-1.0.typelib
%{_libdir}/girepository-1.0/GstCodecs-1.0.typelib
%{_libdir}/girepository-1.0/GstCuda-1.0.typelib
%{_libdir}/girepository-1.0/GstDxva-1.0.typelib
%{_libdir}/girepository-1.0/GstInsertBin-1.0.typelib
%{_libdir}/girepository-1.0/GstMpegts-1.0.typelib
%{_libdir}/girepository-1.0/GstMse-1.0.typelib
%{_libdir}/girepository-1.0/GstPlay-1.0.typelib
%{_libdir}/girepository-1.0/GstPlayer-1.0.typelib
%{_libdir}/girepository-1.0/GstTranscoder-1.0.typelib
@ -705,11 +724,14 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%endif
%{_datadir}/gir-1.0/CudaGst-%{majorminor}.gir
%{_datadir}/gir-1.0/GstAnalytics-%{majorminor}.gir
%{_datadir}/gir-1.0/GstBadAudio-%{majorminor}.gir
%{_datadir}/gir-1.0/GstCodecs-%{majorminor}.gir
%{_datadir}/gir-1.0/GstCuda-%{majorminor}.gir
%{_datadir}/gir-1.0/GstDxva-%{majorminor}.gir
%{_datadir}/gir-1.0/GstInsertBin-%{majorminor}.gir
%{_datadir}/gir-1.0/GstMpegts-%{majorminor}.gir
%{_datadir}/gir-1.0/GstMse-%{majorminor}.gir
%{_datadir}/gir-1.0/GstPlay-%{majorminor}.gir
%{_datadir}/gir-1.0/GstPlayer-%{majorminor}.gir
%{_datadir}/gir-1.0/GstTranscoder-%{majorminor}.gir
@ -718,15 +740,18 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_datadir}/gir-1.0/GstVulkanWayland-%{majorminor}.gir
%{_datadir}/gir-1.0/GstWebRTC-%{majorminor}.gir
%{_libdir}/libgstanalytics-%{majorminor}.so
%{_libdir}/libgstadaptivedemux-%{majorminor}.so
%{_libdir}/libgstbasecamerabinsrc-%{majorminor}.so
%{_libdir}/libgstbadaudio-%{majorminor}.so
%{_libdir}/libgstcuda-%{majorminor}.so
%{_libdir}/libgstcodecparsers-%{majorminor}.so
%{_libdir}/libgstcodecs-%{majorminor}.so
%{_libdir}/libgstdxva-%{majorminor}.so
%{_libdir}/libgstinsertbin-%{majorminor}.so
%{_libdir}/libgstisoff-%{majorminor}.so
%{_libdir}/libgstmpegts-%{majorminor}.so
%{_libdir}/libgstmse-%{majorminor}.so
#{_libdir}/libgstopencv-%{majorminor}.so
%{_libdir}/libgstplay-%{majorminor}.so
%{_libdir}/libgstplayer-%{majorminor}.so
@ -745,6 +770,7 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%endif
%{_includedir}/gstreamer-%{majorminor}/gst/audio
%{_includedir}/gstreamer-%{majorminor}/gst/analytics
%{_includedir}/gstreamer-%{majorminor}/gst/basecamerabinsrc
%{_includedir}/gstreamer-%{majorminor}/gst/codecparsers
%{_includedir}/gstreamer-%{majorminor}/gst/cuda/
@ -752,6 +778,7 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_includedir}/gstreamer-%{majorminor}/gst/interfaces/photography*
%{_includedir}/gstreamer-%{majorminor}/gst/isoff/
%{_includedir}/gstreamer-%{majorminor}/gst/mpegts
%{_includedir}/gstreamer-%{majorminor}/gst/mse/
#{_includedir}/gstreamer-%{majorminor}/gst/opencv
%{_includedir}/gstreamer-%{majorminor}/gst/play
%{_includedir}/gstreamer-%{majorminor}/gst/player
@ -764,11 +791,13 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%{_includedir}/gstreamer-%{majorminor}/gst/webrtc/
# pkg-config files
%{_libdir}/pkgconfig/gstreamer-analytics-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-bad-audio-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-cuda-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-codecparsers-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-insertbin-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-mpegts-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-mse-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-photography-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-play-%{majorminor}.pc
%{_libdir}/pkgconfig/gstreamer-player-%{majorminor}.pc
@ -786,15 +815,35 @@ rm $RPM_BUILD_ROOT%{_bindir}/playout
%changelog
* Mon Jul 15 2024 Wim Taymans <wtaymans@redhat.com> - 1.22.12-3
- Add patch to use new webrtc version
- Resolves: RHEL-28928
* Mon Jul 29 2024 Gwyn Ciesla <gwync@protonmail.com> - 1.24.6-1
- 1.24.6
* Mon Jun 24 2024 Troy Dawson <tdawson@redhat.com> - 1.22.12-2
- Bump release for June 2024 mass rebuild
* Thu Jul 25 2024 Sérgio Basto <sergio@serjux.com> - 1.24.5-3
- Rebuild for opencv 4.10.0
* Fri Jun 14 2024 Wim Taymans <wtaymans@redhat.com> - 1.22.12-1
- Update to 1.22.12
* Thu Jul 18 2024 Fedora Release Engineering <releng@fedoraproject.org> - 1.24.5-2
- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild
* Fri Jun 21 2024 Gwyn Ciesla <gwync@protonmail.com> - 1.24.5-1
- 1.24.5
* Thu May 30 2024 Robert-André Mauchin <zebob.m@gmail.com> - 1.24.4-2
- Rebuild for svt-av1 2.1.0
* Wed May 29 2024 Gwyn Ciesla <gwync@protonmail.com> - 1.24.4-1
- 1.24.4
* Tue Apr 30 2024 Gwyn Ciesla <gwync@protonmail.com> - 1.24.3-1
- 1.24.3
* Mon Apr 22 2024 Gwyn Ciesla <gwync@protonmail.com> - 1.24.0-3
- openexr rebuild
* Wed Mar 13 2024 Yaakov Selkowitz <yselkowi@redhat.com> - 1.24.0-2
- Re-enable webrtcdsp for f40+ and ELN
* Tue Mar 05 2024 Wim Taymans <wtaymans@redhat.com> - 1.24.0-1
- Update to 1.24.0
* Thu Feb 08 2024 Kalev Lember <klember@redhat.com> - 1.22.9-3
- Add gstreamer1-plugin-openh264 subpackage with the openh264 plugin

149
openh264-drop-runtime-version-checks.patch
View File

@ -1,149 +0,0 @@
From 1dadccd48c97a4b7c96ae0307c2263107e7f1876 Mon Sep 17 00:00:00 2001
From: Kalev Lember <klember@redhat.com>
Date: Wed, 6 Dec 2023 14:58:38 +0100
Subject: [PATCH] openh264: Drop runtime version checks
With the way the runtime checks are currently set up, every single
openh264 release, no matter how minor, is considered an ABI break and
requires gst-plugins-bad recompilation. This is unnecessarily strict
because it doesn't allow downstream distributions to ship any openh264
bug fix version updates without breaking gstreamer's openh264 support.
Years ago, at the time when gstreamer's openh264 support was merged,
openh264 releases were done without a versioned soname (the library was
just libopenh264.so, unversioned). Since then, starting with version
1.3.0, openh264 has started using versioned sonames and the intent has
been to bump the soname every time there's a new release with an ABI
change.
This patch drops the strict version check. meson.build already has a
minimum requirement on openh264 version 1.3.0 where soname versioning
was added, which should be good enough to ensure that the library is
using soname versioning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5780>
---
.../ext/openh264/gstopenh264dec.cpp | 7 +--
.../ext/openh264/gstopenh264element.c | 48 -------------------
.../ext/openh264/gstopenh264elements.h | 2 -
.../ext/openh264/gstopenh264enc.cpp | 7 +--
.../gst-plugins-bad/ext/openh264/meson.build | 1 -
5 files changed, 4 insertions(+), 61 deletions(-)
delete mode 100644 subprojects/gst-plugins-bad/ext/openh264/gstopenh264element.c
diff --git a/subprojects/gst-plugins-bad/ext/openh264/gstopenh264dec.cpp b/subprojects/gst-plugins-bad/ext/openh264/gstopenh264dec.cpp
index 77f2b8fe348..f3302567c7b 100644
--- a/subprojects/gst-plugins-bad/ext/openh264/gstopenh264dec.cpp
+++ b/subprojects/gst-plugins-bad/ext/openh264/gstopenh264dec.cpp
@@ -459,10 +459,7 @@ openh264dec_element_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (gst_openh264dec_debug_category, "openh264dec", 0,
"debug category for openh264dec element");
- if (openh264_element_init (plugin))
- return gst_element_register (plugin, "openh264dec", GST_RANK_MARGINAL,
- GST_TYPE_OPENH264DEC);
- GST_ERROR ("Incorrect library version loaded, expecting %s", g_strCodecVer);
- return FALSE;
+ return gst_element_register (plugin, "openh264dec", GST_RANK_MARGINAL,
+ GST_TYPE_OPENH264DEC);
}
diff --git a/subprojects/gst-plugins-bad/ext/openh264/gstopenh264element.c b/subprojects/gst-plugins-bad/ext/openh264/gstopenh264element.c
deleted file mode 100644
index 3c5c378c81e..00000000000
--- a/subprojects/gst-plugins-bad/ext/openh264/gstopenh264element.c
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2014, Ericsson AB. All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without modification,
- * are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice, this
- * list of conditions and the following disclaimer.
- *
- * 2. Redistributions in binary form must reproduce the above copyright notice, this
- * list of conditions and the following disclaimer in the documentation and/or other
- * materials provided with the distribution.
- *
- * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
- * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
- * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
- * IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
- * INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
- * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
- * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY
- * OF SUCH DAMAGE.
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <gst/gst.h>
-#include <wels/codec_api.h>
-#include <wels/codec_ver.h>
-#include <string.h>
-#include "gstopenh264elements.h"
-
-
-gboolean
-openh264_element_init (GstPlugin * plugin)
-{
- OpenH264Version libver;
- /* g_stCodecVersion is the version detected at build time as defined in the
- * headers and WelsGetCodecVersion() is the version detected at runtime.
- * This is a safeguard to avoid crashes since OpenH264 has been changing
- * ABI without changing the SONAME.
- */
- libver = WelsGetCodecVersion ();
- return (memcmp (&libver, &g_stCodecVersion, sizeof (libver)) == 0);
-}
diff --git a/subprojects/gst-plugins-bad/ext/openh264/gstopenh264elements.h b/subprojects/gst-plugins-bad/ext/openh264/gstopenh264elements.h
index 572f6a8e078..5c9582941ee 100644
--- a/subprojects/gst-plugins-bad/ext/openh264/gstopenh264elements.h
+++ b/subprojects/gst-plugins-bad/ext/openh264/gstopenh264elements.h
@@ -27,8 +27,6 @@
G_BEGIN_DECLS
-gboolean openh264_element_init (GstPlugin * plugin);
-
GST_ELEMENT_REGISTER_DECLARE (openh264dec);
GST_ELEMENT_REGISTER_DECLARE (openh264enc);
diff --git a/subprojects/gst-plugins-bad/ext/openh264/gstopenh264enc.cpp b/subprojects/gst-plugins-bad/ext/openh264/gstopenh264enc.cpp
index 6b54b1584f8..05c126cfc64 100644
--- a/subprojects/gst-plugins-bad/ext/openh264/gstopenh264enc.cpp
+++ b/subprojects/gst-plugins-bad/ext/openh264/gstopenh264enc.cpp
@@ -1066,10 +1066,7 @@ openh264enc_element_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (gst_openh264enc_debug_category, "openh264enc", 0,
"debug category for openh264enc element");
- if (openh264_element_init (plugin))
- return gst_element_register (plugin, "openh264enc", GST_RANK_MARGINAL,
- GST_TYPE_OPENH264ENC);
- GST_ERROR ("Incorrect library version loaded, expecting %s", g_strCodecVer);
- return FALSE;
+ return gst_element_register (plugin, "openh264enc", GST_RANK_MARGINAL,
+ GST_TYPE_OPENH264ENC);
}
diff --git a/subprojects/gst-plugins-bad/ext/openh264/meson.build b/subprojects/gst-plugins-bad/ext/openh264/meson.build
index 1f0a198b05e..c6f247e1cdd 100644
--- a/subprojects/gst-plugins-bad/ext/openh264/meson.build
+++ b/subprojects/gst-plugins-bad/ext/openh264/meson.build
@@ -1,7 +1,6 @@
openh264_sources = [
'gstopenh264dec.cpp',
'gstopenh264enc.cpp',
- 'gstopenh264element.c',
'gstopenh264plugin.c',
]
--
GitLab

2
sources
View File

@ -1 +1 @@
SHA512 (gst-plugins-bad-free-1.22.12.tar.xz) = 791a35edcfbf3ac6e4442ff0cf43f41edd547ebeeafa33771c365345dd04319453bea7a0a3d706a665e7e5c579d45072b182cfecae5595c25064653745e96972
SHA512 (gst-plugins-bad-free-1.24.6.tar.xz) = 174f3d0ee0ff8e6e73768ddac2fe82cd250c1295f64dddf87c9bf33d175422baf2a05989f019464463bb955920f639c3c651c56f32c0e7587731425ae8aca3e7