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Disable analyzer for other bit depths than 16

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This removes the splitting of the pipeline with the tee.
Move HandoffCallback to the source, which makes it possible to convert the audio buffer in HandoffCallback later.
Until then just disable analyzer for other formats.

Removes tee and probe queue converter and sink
This commit is contained in:
Jonas Kvinge
2019-10-19 01:45:24 +02:00
parent 4d888dfce8
commit abeb580228
4 changed files with 35 additions and 55 deletions
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4
src/engine/gstbufferconsumer.h
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@@ -25,6 +25,8 @@
#include <gst/gstbuffer.h>
#include <QString>
class GstEnginePipeline;
class GstBufferConsumer {
@@ -33,7 +35,7 @@ public:
// This is called in some unspecified GStreamer thread.
// Ownership of the buffer is transferred to the BufferConsumer and it should gst_buffer_unref it.
virtual void ConsumeBuffer(GstBuffer *buffer, int pipeline_id) = 0;
virtual void ConsumeBuffer(GstBuffer *buffer, const int pipeline_id, const QString &format) = 0;
};
#endif // GSTBUFFERCONSUMER_H

19
src/engine/gstengine.cpp
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@@ -411,10 +411,10 @@ GstElement *GstEngine::CreateElement(const QString &factoryName, GstElement *bin
return element;
}
void GstEngine::ConsumeBuffer(GstBuffer *buffer, const int pipeline_id) {
void GstEngine::ConsumeBuffer(GstBuffer *buffer, const int pipeline_id, const QString &format) {
// Schedule this to run in the GUI thread. The buffer gets added to the queue and unreffed by UpdateScope.
if (!QMetaObject::invokeMethod(this, "AddBufferToScope", Q_ARG(GstBuffer*, buffer), Q_ARG(int, pipeline_id))) {
if (!QMetaObject::invokeMethod(this, "AddBufferToScope", Q_ARG(GstBuffer*, buffer), Q_ARG(int, pipeline_id), Q_ARG(QString, format))) {
qLog(Warning) << "Failed to invoke AddBufferToScope on GstEngine";
}
@@ -520,7 +520,7 @@ void GstEngine::NewMetaData(const int pipeline_id, const Engine::SimpleMetaBundl
}
void GstEngine::AddBufferToScope(GstBuffer *buf, const int pipeline_id) {
void GstEngine::AddBufferToScope(GstBuffer *buf, const int pipeline_id, const QString &format) {
if (!current_pipeline_ || current_pipeline_->id() != pipeline_id) {
gst_buffer_unref(buf);
@@ -531,6 +531,7 @@ void GstEngine::AddBufferToScope(GstBuffer *buf, const int pipeline_id) {
gst_buffer_unref(latest_buffer_);
}
buffer_format_ = format;
latest_buffer_ = buf;
have_new_buffer_ = true;
@@ -806,13 +807,23 @@ void GstEngine::UpdateScope(const int chunk_length) {
}
scope_chunk_++;
memcpy(dest, source, bytes);
if (buffer_format_ == "S16LE" ||
buffer_format_ == "S16BE" ||
buffer_format_ == "U16LE" ||
buffer_format_ == "U16BE" ||
buffer_format_ == "S16" ||
buffer_format_ == "U16")
memcpy(dest, source, bytes);
else
memset(dest, 0, bytes);
gst_buffer_unmap(latest_buffer_, &map);
if (scope_chunk_ == scope_chunks_) {
gst_buffer_unref(latest_buffer_);
latest_buffer_ = nullptr;
buffer_format_.clear();
}
}

5
src/engine/gstengine.h
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@@ -95,7 +95,7 @@ class GstEngine : public Engine::Base, public GstBufferConsumer {
void EnsureInitialised() { gst_startup_->EnsureInitialised(); }
GstElement *CreateElement(const QString &factoryName, GstElement *bin = nullptr, const bool showerror = true);
void ConsumeBuffer(GstBuffer *buffer, int pipeline_id);
void ConsumeBuffer(GstBuffer *buffer, const int pipeline_id, const QString &format);
public slots:
@@ -124,7 +124,7 @@ class GstEngine : public Engine::Base, public GstBufferConsumer {
void EndOfStreamReached(const int pipeline_id, const bool has_next_track);
void HandlePipelineError(const int pipeline_id, const QString &message, const int domain, const int error_code);
void NewMetaData(const int pipeline_id, const Engine::SimpleMetaBundle &bundle);
void AddBufferToScope(GstBuffer *buf, const int pipeline_id);
void AddBufferToScope(GstBuffer *buf, const int pipeline_id, const QString &format);
void FadeoutFinished();
void FadeoutPauseFinished();
void SeekNow();
@@ -201,6 +201,7 @@ class GstEngine : public Engine::Base, public GstBufferConsumer {
int scope_chunk_;
bool have_new_buffer_;
int scope_chunks_;
QString buffer_format_;
#ifdef Q_OS_MACOS
GTlsDatabase* tls_database_;

62
src/engine/gstenginepipeline.cpp
View File

@@ -226,18 +226,6 @@ GstElement *GstEnginePipeline::CreateDecodeBinFromString(const char *pipeline) {
bool GstEnginePipeline::InitAudioBin() {
// Here we create all the parts of the gstreamer pipeline - from the source to the sink. The parts of the pipeline are split up into bins:
// uri decode bin -> audio bin
// The uri decode bin is a gstreamer builtin that automatically picks the right type of source and decoder for the URI.
// The audio bin gets created here and contains:
// queue ! audioconvert ! <caps32> ! ( rgvolume ! rglimiter ! audioconvert2 ) ! tee
// rgvolume and rglimiter are only created when replaygain is enabled.
// After the tee the pipeline splits. One split is converted to 16-bit int samples for the scope, the other is kept as float32 and sent to the speaker.
// tee1 ! probe_queue ! probe_converter ! <caps16> ! probe_sink
// tee2 ! audio_queue ! equalizer_preamp ! equalizer ! volume ! audioscale ! convert ! audiosink
gst_segment_init(&last_decodebin_segment_, GST_FORMAT_TIME);
// Audio bin
@@ -276,10 +264,6 @@ bool GstEnginePipeline::InitAudioBin() {
queue_ = engine_->CreateElement("queue2", audiobin_);
audioconvert_ = engine_->CreateElement("audioconvert", audiobin_);
GstElement *tee = engine_->CreateElement("tee", audiobin_);
GstElement *probe_queue = engine_->CreateElement("queue2", audiobin_);
GstElement *probe_converter = engine_->CreateElement("audioconvert", audiobin_);
GstElement *probe_sink = engine_->CreateElement("fakesink", audiobin_);
GstElement *audio_queue = engine_->CreateElement("queue", audiobin_);
audioscale_ = engine_->CreateElement("audioresample", audiobin_);
GstElement *convert = engine_->CreateElement("audioconvert", audiobin_);
@@ -294,7 +278,7 @@ bool GstEnginePipeline::InitAudioBin() {
equalizer_ = engine_->CreateElement("equalizer-nbands", audiobin_, false);
}
if (!queue_ || !audioconvert_ || !tee || !probe_queue || !probe_converter || !probe_sink || !audio_queue || !audioscale_ || !convert) {
if (!queue_ || !audioconvert_ || !audio_queue || !audioscale_ || !convert) {
gst_object_unref(GST_OBJECT(audiobin_));
audiobin_ = nullptr;
return false;
@@ -304,7 +288,7 @@ bool GstEnginePipeline::InitAudioBin() {
// event_probe is the audioconvert element we attach the probe to, which will change depending on whether replaygain is enabled.
// convert_sink is the element after the first audioconvert, which again will change.
GstElement *event_probe = audioconvert_;
GstElement *convert_sink = tee;
GstElement *convert_sink = audio_queue;
if (rg_enabled_) {
rgvolume_ = engine_->CreateElement("rgvolume", audiobin_, false);
@@ -331,8 +315,9 @@ bool GstEnginePipeline::InitAudioBin() {
gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_EVENT_UPSTREAM, &EventHandoffCallback, this, nullptr);
gst_object_unref(pad);
// Configure the fakesink properly
g_object_set(G_OBJECT(probe_sink), "sync", TRUE, nullptr);
pad = gst_element_get_static_pad(queue_, "sink");
gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, HandoffCallback, this, nullptr);
gst_object_unref(pad);
// Setting the equalizer bands:
//
@@ -371,7 +356,8 @@ bool GstEnginePipeline::InitAudioBin() {
// Set the stereo balance.
if (audio_panorama_) g_object_set(G_OBJECT(audio_panorama_), "panorama", stereo_balance_, nullptr);
// Set the buffer duration. We set this on this queue instead of the decode bin (in ReplaceDecodeBin()) because setting it on the decode bin only affects network sources.
// Set the buffer duration.
// We set this on this queue instead of the playbin because setting it on the playbin only affects network sources.
// Disable the default buffer and byte limits, so we only buffer based on time.
g_object_set(G_OBJECT(queue_), "max-size-buffers", 0, nullptr);
@@ -382,32 +368,13 @@ bool GstEnginePipeline::InitAudioBin() {
g_object_set(G_OBJECT(queue_), "use-buffering", true, nullptr);
}
g_object_set(G_OBJECT(probe_queue), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(probe_queue), "max-size-bytes", 0, nullptr);
g_object_set(G_OBJECT(probe_queue), "max-size-time", 0, nullptr);
g_object_set(G_OBJECT(probe_queue), "low-watermark", 0.0, nullptr);
g_object_set(G_OBJECT(probe_queue), "use-buffering", true, nullptr);
gst_element_link_many(queue_, audioconvert_, convert_sink, nullptr);
gst_element_link(probe_converter, probe_sink);
// Link the outputs of tee to the queues on each path.
pad = gst_element_get_static_pad(probe_queue, "sink");
gst_pad_link(gst_element_get_request_pad(tee, "src_%u"), pad);
gst_object_unref(pad);
pad = gst_element_get_static_pad(audio_queue, "sink");
gst_pad_link(gst_element_get_request_pad(tee, "src_%u"), pad);
gst_object_unref(pad);
// Link replaygain elements if enabled.
if (rg_enabled_ && rgvolume_ && rglimiter_ && audioconvert2_) {
gst_element_link_many(rgvolume_, rglimiter_, audioconvert2_, tee, nullptr);
gst_element_link_many(rgvolume_, rglimiter_, audioconvert2_, audio_queue, nullptr);
}
// Don't force 16 bit.
gst_element_link(probe_queue, probe_converter);
if (eq_enabled_ && equalizer_ && equalizer_preamp_ && audio_panorama_) {
if (volume_)
gst_element_link_many(audio_queue, equalizer_preamp_, equalizer_, audio_panorama_, volume_, audioscale_, convert, nullptr);
@@ -424,11 +391,6 @@ bool GstEnginePipeline::InitAudioBin() {
gst_element_link_filtered(convert, audiosink_, caps);
gst_caps_unref(caps);
// Add probes and handlers.
pad = gst_element_get_static_pad(probe_converter, "src");
gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, HandoffCallback, this, nullptr);
gst_object_unref(pad);
GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline_));
gst_bus_set_sync_handler(bus, BusCallbackSync, this, nullptr);
bus_cb_id_ = gst_bus_add_watch(bus, BusCallback, this);
@@ -849,11 +811,15 @@ GstPadProbeReturn GstEnginePipeline::DecodebinProbe(GstPad *pad, GstPadProbeInfo
}
GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad*, GstPadProbeInfo *info, gpointer self) {
GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad *pad, GstPadProbeInfo *info, gpointer self) {
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
if (!instance) return GST_PAD_PROBE_OK;
GstCaps *caps = gst_pad_get_current_caps(pad);
GstStructure *structure = gst_caps_get_structure(caps, 0);
const gchar *format = gst_structure_get_string(structure, "format");
GstBuffer *buf = gst_pad_probe_info_get_buffer(info);
QList<GstBufferConsumer*> consumers;
@@ -864,7 +830,7 @@ GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad*, GstPadProbeInfo *i
for (GstBufferConsumer *consumer : consumers) {
gst_buffer_ref(buf);
consumer->ConsumeBuffer(buf, instance->id());
consumer->ConsumeBuffer(buf, instance->id(), QString(format));
}
// Calculate the end time of this buffer so we can stop playback if it's after the end time of this song.