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Remove the precise analyzer timing code from Amarok and instead just use the

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latest buffer from GStreamer.  This works just as well and fixes the occasional
freezing analyzer.  Fixes issue 2464
This commit is contained in:
David Sansome 2013-06-01 18:03:59 +10:00
parent 90898bd255
commit 619261f5e1
2 changed files with 23 additions and 137 deletions
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154
src/engines/gstengine.cpp
View File

@ -70,8 +70,7 @@ GstEngine::GstEngine(TaskManager* task_manager)
: Engine::Base(),
task_manager_(task_manager),
buffering_task_id_(-1),
delayq_(g_queue_new()),
current_sample_(0),
latest_buffer_(NULL),
equalizer_enabled_(false),
stereo_balance_(0.0f),
rg_enabled_(false),
@ -98,10 +97,6 @@ GstEngine::~GstEngine() {
current_pipeline_.reset();
// Destroy scope delay queue
ClearScopeBuffers();
g_queue_free(delayq_);
// Save configuration
gst_deinit();
}
@ -187,97 +182,44 @@ void GstEngine::AddBufferToScope(GstBuffer* buf, int pipeline_id) {
return;
}
g_queue_push_tail(delayq_, buf);
if (latest_buffer_ != NULL) {
gst_buffer_unref(latest_buffer_);
}
latest_buffer_ = buf;
}
const Engine::Scope& GstEngine::scope() {
UpdateScope();
if (current_sample_ >= kScopeSize) {
// ok, we have a full buffer now, so give it to the scope
for (int i=0; i< kScopeSize; i++)
scope_[i] = current_scope_[i];
current_sample_ = 0;
if (latest_buffer_ != NULL) {
UpdateScope();
}
return scope_;
}
void GstEngine::UpdateScope() {
typedef int16_t sampletype;
// prune the scope and get the current pos of the audio device
const quint64 pos = PruneScope();
const quint64 segment_start = current_pipeline_->segment_start();
// head of the delay queue is the most delayed, so we work with that one
GstBuffer *buf = reinterpret_cast<GstBuffer *>( g_queue_peek_head(delayq_) );
if (!buf)
return;
// start time for this buffer
quint64 stime = GST_BUFFER_TIMESTAMP(buf) - segment_start;
// duration of the buffer...
quint64 dur = GST_BUFFER_DURATION(buf);
// therefore we can calculate the end time for the buffer
quint64 etime = stime + dur;
typedef typename Engine::Scope::value_type sample_type;
// determine the number of channels
GstStructure* structure = gst_caps_get_structure ( GST_BUFFER_CAPS( buf ), 0);
GstStructure* structure = gst_caps_get_structure(
GST_BUFFER_CAPS(latest_buffer_), 0);
int channels = 2;
gst_structure_get_int (structure, "channels", &channels);
gst_structure_get_int(structure, "channels", &channels);
// scope does not support >2 channels
if (channels > 2)
return;
// if the audio device is playing this buffer now
if (pos <= stime || pos >= etime)
return;
const sample_type* source = reinterpret_cast<sample_type*>(
GST_BUFFER_DATA(latest_buffer_));
sample_type* dest = scope_.data();
const int bytes = qMin(
static_cast<Engine::Scope::size_type>(GST_BUFFER_SIZE(latest_buffer_)),
scope_.size() * sizeof(sample_type));
// calculate the number of samples in the buffer
int sz = GST_BUFFER_SIZE(buf) / sizeof(sampletype);
// number of frames is the number of samples in each channel (frames like in the alsa sense)
int frames = sz / channels;
memcpy(dest, source, bytes);
// find the offset into the buffer to the sample closest to where the audio device is playing
// it is the (time into the buffer cooresponding to the audio device pos) / (the sample rate)
// sample rate = duration of the buffer / number of frames in the buffer
// then we multiply by the number of channels to find the offset of the left channel sample
// of the frame in the buffer
int off = channels * (pos - stime) / (dur / frames);
// note that we are assuming 32 bit samples, but this should probably be generalized...
sampletype* data = reinterpret_cast<sampletype *>(GST_BUFFER_DATA(buf));
if (off >= sz) // better be...
return;
int i = off; // starting at offset
// loop while we fill the current buffer. If we need another buffer and one is available,
// get it and keep filling. If there are no more buffers available (not too likely)
// then leave everything in this state and wait until the next time the scope updates
while (buf && current_sample_ < kScopeSize && i < sz) {
for (int j = 0; j < channels && current_sample_ < kScopeSize; j++) {
current_scope_[current_sample_ ++] = data[i + j];
}
i+=channels; // advance to the next frame
if (i >= sz - 1) {
// here we are out of samples in the current buffer, so we get another one
buf = reinterpret_cast<GstBuffer *>( g_queue_pop_head(delayq_) );
gst_buffer_unref(buf);
buf = reinterpret_cast<GstBuffer *>( g_queue_peek_head(delayq_) );
if (buf) {
stime = GST_BUFFER_TIMESTAMP(buf);
dur = GST_BUFFER_DURATION(buf);
etime = stime + dur;
i = 0;
sz = GST_BUFFER_SIZE(buf) / sizeof(sampletype);
data = reinterpret_cast<sampletype *>(GST_BUFFER_DATA(buf));
}
}
}
gst_buffer_unref(latest_buffer_);
latest_buffer_ = NULL;
}
void GstEngine::StartPreloading(const QUrl& url, bool force_stop_at_end,
@ -360,7 +302,6 @@ void GstEngine::StartFadeout() {
fadeout_pipeline_ = current_pipeline_;
disconnect(fadeout_pipeline_.get(), 0, 0, 0);
fadeout_pipeline_->RemoveAllBufferConsumers();
ClearScopeBuffers();
fadeout_pipeline_->StartFader(fadeout_duration_nanosec_, QTimeLine::Backward);
connect(fadeout_pipeline_.get(), SIGNAL(FaderFinished()), SLOT(FadeoutFinished()));
@ -433,8 +374,6 @@ void GstEngine::PlayDone() {
StartTimers();
current_sample_ = 0;
// initial offset
if(offset_nanosec != 0 || beginning_nanosec_ != 0) {
Seek(offset_nanosec);
@ -553,10 +492,9 @@ void GstEngine::SeekNow() {
if (!current_pipeline_)
return;
if (current_pipeline_->Seek(seek_pos_))
ClearScopeBuffers();
else
if (!current_pipeline_->Seek(seek_pos_)) {
qLog(Warning) << "Seek failed";
}
}
void GstEngine::SetEqualizerEnabled(bool enabled) {
@ -604,11 +542,6 @@ void GstEngine::timerEvent(QTimerEvent* e) {
if (e->timerId() != timer_id_)
return;
// keep the scope from building while we are not visible
// this is why the timer must run as long as we are playing, and not just when
// we are fading
PruneScope();
if (current_pipeline_) {
const qint64 current_position = position_nanosec();
const qint64 current_length = length_nanosec();
@ -666,7 +599,6 @@ void GstEngine::EndOfStreamReached(int pipeline_id, bool has_next_track) {
current_pipeline_.reset();
BufferingFinished();
}
ClearScopeBuffers();
emit TrackEnded();
}
@ -772,48 +704,6 @@ shared_ptr<GstEnginePipeline> GstEngine::CreatePipeline(const QUrl& url,
return ret;
}
qint64 GstEngine::PruneScope() {
if (!current_pipeline_)
return 0;
// get the position playing in the audio device
const qint64 pos = current_pipeline_->position();
const qint64 segment_start = current_pipeline_->segment_start();
GstBuffer *buf = 0;
quint64 etime = 0;
// free up the buffers that the audio device has advanced past already
do {
// most delayed buffers are at the head of the queue
buf = reinterpret_cast<GstBuffer *>( g_queue_peek_head(delayq_) );
if (buf) {
// the start time of the buffer
quint64 stime = GST_BUFFER_TIMESTAMP(buf) - segment_start;
// the duration of the buffer
quint64 dur = GST_BUFFER_DURATION(buf);
// therefore we can calculate the end time of the buffer
etime = stime + dur;
// purge this buffer if the pos is past the end time of the buffer
if (pos > qint64(etime)) {
g_queue_pop_head(delayq_);
gst_buffer_unref(buf);
}
}
} while (buf && pos > qint64(etime));
return pos;
}
void GstEngine::ClearScopeBuffers() {
// just free them all
while (g_queue_get_length(delayq_)) {
GstBuffer* buf = reinterpret_cast<GstBuffer *>( g_queue_pop_head(delayq_) );
gst_buffer_unref(buf);
}
}
bool GstEngine::DoesThisSinkSupportChangingTheOutputDeviceToAUserEditableString(const QString &name) {
return (name == "alsasink" || name == "osssink" || name == "pulsesink");
}

6
src/engines/gstengine.h
View File

@ -121,7 +121,6 @@ class GstEngine : public Engine::Base, public BufferConsumer {
void EndOfStreamReached(int pipeline_id, bool has_next_track);
void HandlePipelineError(int pipeline_id, const QString& message, int domain, int error_code);
void NewMetaData(int pipeline_id, const Engine::SimpleMetaBundle& bundle);
void ClearScopeBuffers();
void AddBufferToScope(GstBuffer* buf, int pipeline_id);
void FadeoutFinished();
void FadeoutPauseFinished();
@ -151,7 +150,6 @@ class GstEngine : public Engine::Base, public BufferConsumer {
boost::shared_ptr<GstEnginePipeline> CreatePipeline(const QUrl& url, qint64 end_nanosec);
void UpdateScope();
qint64 PruneScope();
int AddBackgroundStream(boost::shared_ptr<GstEnginePipeline> pipeline);
@ -180,9 +178,7 @@ class GstEngine : public Engine::Base, public BufferConsumer {
QList<BufferConsumer*> buffer_consumers_;
GQueue* delayq_;
float current_scope_[kScopeSize];
int current_sample_;
GstBuffer* latest_buffer_;
bool equalizer_enabled_;
int equalizer_preamp_;