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gstaamp.cpp
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gstaamp.cpp
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/*
* Copyright 2018 RDK Management
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation, version 2
* of the license.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* @file gstaamp.cpp
* @brief AAMP gstreamer plugin definitions
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <string.h>
#include <stdio.h>
#include "gstaamp.h"
#include "main_aamp.h"
#include "priv_aamp.h"
GST_DEBUG_CATEGORY_STATIC (gst_aamp_debug_category);
#define GST_CAT_DEFAULT gst_aamp_debug_category
#define MAX_BYTES_TO_SEND (188*1024)
#define MAX_NUM_BUFFERS_IN_QUEUE 30
#define GST_AAMP_LOG_TIMING(msg...) GST_FIXME_OBJECT(aamp, msg)
#define STREAM_COUNT (sizeof(aamp->stream)/sizeof(aamp->stream[0]))
static const gchar *g_aamp_expose_hls_caps = NULL;
static GstStateChangeReturn
gst_aamp_change_state(GstElement * element, GstStateChange transition);
static void gst_aamp_finalize(GObject * object);
static gboolean gst_aamp_query(GstElement * element, GstQuery * query);
static GstFlowReturn gst_aamp_sink_chain(GstPad * pad, GstObject *parent, GstBuffer * buffer);
static gboolean gst_aamp_sink_event(GstPad * pad, GstObject *parent, GstEvent * event);
static gboolean gst_aamp_src_event(GstPad * pad, GstObject *parent, GstEvent * event);
static gboolean gst_aamp_src_query(GstPad * pad, GstObject *parent, GstQuery * query);
static void gst_aamp_configure(GstAamp * aamp, StreamOutputFormat format, StreamOutputFormat audioFormat);
static gboolean gst_aamp_ready(GstAamp *aamp);
#ifdef AAMP_JSCONTROLLER_ENABLED
extern "C"
{
void setAAMPPlayerInstance(PlayerInstanceAAMP *, int);
void unsetAAMPPlayerInstance(PlayerInstanceAAMP *);
}
#endif
/**
* @enum GstAampProperties
* @brief Placeholder for gstaamp properties
*/
enum GstAampProperties
{
PROP_0
};
gboolean gst_aamp_push(media_stream* stream, GstMiniObject *obj, gboolean *eosEvent = NULL)
{
GST_TRACE_OBJECT(stream->parent, "Enter gst_aamp_push");
gboolean retVal = TRUE;
if (GST_IS_BUFFER(obj))
{
if (stream->isPaused)
{
GST_WARNING_OBJECT(stream->parent, "gst_pad_push[%s] paused\n", GST_PAD_NAME(stream->srcpad));
return FALSE;
}
GstFlowReturn ret;
ret = gst_pad_push(stream->srcpad, GST_BUFFER(obj));
if (ret != GST_FLOW_OK)
{
GST_WARNING_OBJECT(stream->parent, "gst_pad_push[%s] error: %s \n", GST_PAD_NAME(stream->srcpad),
gst_flow_get_name(ret));
retVal = gst_pad_pause_task(stream->srcpad);
if (!retVal)
GST_WARNING_OBJECT(stream->parent, "gst_pad_push[%s] pausing error \n", GST_PAD_NAME(stream->srcpad));
stream->isPaused=TRUE;
retVal = FALSE;
}
}
else if (GST_IS_EVENT(obj))
{
GstEvent* event = GST_EVENT(obj);
if (eosEvent)
{
if (GST_EVENT_TYPE(event) == GST_EVENT_EOS)
{
*eosEvent = TRUE;
}
}
GST_INFO_OBJECT(stream->parent, "%s: send %s event\n", __FUNCTION__,
GST_EVENT_TYPE_NAME(event));
if (!gst_pad_push_event(stream->srcpad, event))
{
GST_WARNING_OBJECT(stream->parent, "gst_pad_push_event[%s] error\n", GST_PAD_NAME(stream->srcpad));
}
}
else
{
GST_ERROR_OBJECT(stream->parent, "%s: unexpected object type in queue\n", __FUNCTION__);
}
return retVal;
}
/**
* @brief Enqueue buffer/event to queue
* @param[in] stream Media stream object pointer
* @param[in] item pointer to the object to be added to queue/pushed
*/
void gst_aamp_stream_add_item(media_stream* stream, gpointer item)
{
GST_TRACE_OBJECT(stream->parent, "Enter gst_aamp_stream_add_item");
GstAamp *aamp = GST_AAMP(stream->parent);
g_assert(NULL != stream->srcpad);
if (stream->parent->enable_src_tasks)
{
g_mutex_lock(&stream->mutex);
while (g_queue_get_length(stream->queue) > MAX_NUM_BUFFERS_IN_QUEUE)
{
g_cond_wait(&stream->cond, &stream->mutex);
if (aamp->flushing)
{
break;
}
}
if (aamp->flushing)
{
GstMiniObject *obj = (GstMiniObject *) item;
if (GST_IS_BUFFER(obj))
{
gst_buffer_unref(GST_BUFFER(obj));
}
else if (GST_IS_EVENT(obj))
{
gst_event_unref(GST_EVENT(obj));
}
else
{
GST_ERROR_OBJECT(aamp, "%s: unexpected object type in queue\n", __FUNCTION__);
}
}
else
{
g_queue_push_tail(stream->queue, item);
}
g_cond_broadcast(&stream->cond);
g_mutex_unlock(&stream->mutex);
}
else
{
gst_aamp_push(stream, (GstMiniObject *) item);
}
}
/**
* @brief Dequeue buffer/event and push it to srcpad
* @param[in] stream Media stream object pointer
*/
void gst_aamp_stream_push_next_item(media_stream* stream)
{
GST_TRACE_OBJECT(stream->parent, "Enter gst_aamp_stream_push_next_item \n");
GstAamp *aamp = GST_AAMP(stream->parent);
gboolean eosSent = FALSE;
while (!eosSent)
{
g_mutex_lock(&stream->mutex);
if (g_queue_is_empty(stream->queue) && !aamp->flushing)
{
g_cond_wait(&stream->cond, &stream->mutex);
}
if (aamp->flushing)
{
GST_INFO_OBJECT(aamp, "Flushing");
g_mutex_unlock(&stream->mutex);
gst_pad_pause_task(stream->srcpad);
break;
}
gpointer item = g_queue_pop_head(stream->queue);
g_cond_broadcast(&stream->cond);
g_mutex_unlock(&stream->mutex);
if (item)
{
if (!gst_aamp_push(stream, (GstMiniObject *)item, &eosSent))
{
break;
}
}
else
{
GST_ERROR_OBJECT(aamp, "%s: object NULL\n", __FUNCTION__);
break;
}
}
}
/**
* @brief Start src pad task of stream
* @param[in] stream Media stream object pointer
*/
void gst_aamp_stream_start(media_stream* stream)
{
GST_DEBUG_OBJECT(stream->parent, "Enter gst_aamp_stream_start");
if(stream->srcpad)
{
GST_INFO_OBJECT(stream->parent, "start %s pad task", GST_PAD_NAME(stream->srcpad));
gst_pad_start_task (stream->srcpad, (GstTaskFunction) gst_aamp_stream_push_next_item,
stream, NULL);
}
}
/**
* @brief Flushes stream queue
* @param[in] aamp Gstreamer aamp object pointer
*/
void gst_aamp_stream_flush(GstAamp *aamp)
{
GST_DEBUG_OBJECT(aamp, "Enter gst_aamp_stream_flush");
for (int i = 0; i < STREAM_COUNT; i++)
{
media_stream* stream = &aamp->stream[i];
if (stream->srcpad)
{
if (aamp->enable_src_tasks)
{
g_mutex_lock(&stream->mutex);
while (FALSE == g_queue_is_empty(stream->queue))
{
GstMiniObject *obj = (GstMiniObject *) g_queue_pop_head(stream->queue);
if (GST_IS_BUFFER(obj))
{
gst_buffer_unref(GST_BUFFER(obj));
}
else if (GST_IS_EVENT(obj))
{
gst_event_unref(GST_EVENT(obj));
}
else
{
GST_ERROR_OBJECT(stream->parent, "%s: unexpected object type in queue\n", __FUNCTION__);
}
}
g_cond_broadcast(&stream->cond);
g_mutex_unlock(&stream->mutex);
}
}
}
}
/**
* @brief Flushes gstreamer elements and stop any running pad tasks
* @param[in] aamp Gstreamer aamp object pointer
*/
void gst_aamp_stop_and_flush(GstAamp *aamp)
{
GST_DEBUG_OBJECT(aamp, "Enter gst_aamp_stop_and_flush");
aamp->flushing = TRUE;
gst_aamp_stream_flush(aamp);
for (int i = 0; i < STREAM_COUNT; i++)
{
media_stream* stream = &aamp->stream[i];
if (stream->srcpad)
{
GST_INFO_OBJECT(aamp, "[%s]sending flush start", GST_PAD_NAME(stream->srcpad));
gst_pad_push_event(stream->srcpad, gst_event_new_flush_start());
if (aamp->enable_src_tasks)
{
gst_pad_stop_task(stream->srcpad);
}
GST_INFO_OBJECT(aamp, "[%s]sending flush stop", GST_PAD_NAME(stream->srcpad));
gst_pad_push_event(stream->srcpad, gst_event_new_flush_stop(TRUE));
}
}
aamp->flushing = FALSE;
}
/**
* @class GstAampStreamer
* @brief Handle media data/configuration/events from AAMP core
*/
class GstAampStreamer : public StreamSink, public AAMPEventObjectListener
{
public:
/**
* @brief GstAampStreamer Constructor
* @param[in] aamp Associated gstaamp pointer
*/
GstAampStreamer(GstAamp * aamp)
{
GST_DEBUG_OBJECT(aamp, "Enter GstAampStreamer");
this->aamp = aamp;
rate = AAMP_NORMAL_PLAY_RATE;
srcPadCapsSent = true;
format = FORMAT_INVALID;
audioFormat = FORMAT_INVALID;
readyToSend = false;
for (int i = 0; i < STREAM_COUNT; i++)
aamp->stream[i].isPaused = FALSE;
aamp->seekFlush = FALSE;
aamp->spts=0.0;
}
/**
* @brief Configures gstaamp with stream output formats
* @param[in] format Output format of main media
* @param[in] audioFormat Output format of audio if present
*/
void Configure(StreamOutputFormat format, StreamOutputFormat audioFormat, bool bESChangeStatus)
{
GST_INFO_OBJECT(aamp, "Enter Configure() format = %d audioFormat = %d, bESChangeStatus %d", format, audioFormat, bESChangeStatus);
this->format = format;
this->audioFormat = audioFormat;
for (int i = 0; i < STREAM_COUNT; i++)
aamp->stream[i].isPaused = FALSE;
aamp->seekFlush = FALSE;
aamp->spts = 0.0;
gst_aamp_configure(aamp, format, audioFormat);
}
/**
* @brief Sends pending events to stream's src pad
* @param[in] stream Media stream to which events are sent
* @param[in] pts Presentation time-stamp
*/
void SendPendingEvents(media_stream* stream, GstClockTime pts)
{
GST_INFO_OBJECT(aamp, "Enter SendPendingEvents");
if (stream->streamStart)
{
GST_INFO_OBJECT(aamp, "sending new_stream_start\n");
gst_aamp_stream_add_item(stream, gst_event_new_stream_start(aamp->stream_id));
GST_INFO_OBJECT(aamp, "%s: sending caps1\n", __FUNCTION__);
gst_aamp_stream_add_item(stream, gst_event_new_caps(stream->caps));
stream->streamStart = FALSE;
GST_INFO_OBJECT(aamp, "%s: sent caps\n", __FUNCTION__);
stream->isPaused=FALSE;
}
if (stream->flush)
{
GST_INFO_OBJECT(aamp, "%s: sending flush start\n", __FUNCTION__);
gst_aamp_stream_add_item(stream, gst_event_new_flush_start());
stream->isPaused=TRUE;
#ifdef USE_GST1
GstEvent* event = gst_event_new_flush_stop(FALSE);
#else
GstEvent* event = gst_event_new_flush_stop();
#endif
GST_INFO_OBJECT(aamp, "%s: sending flush stop\n", __FUNCTION__);
gst_aamp_stream_add_item(stream, event);
stream->flush = FALSE;
}
if (stream->resetPosition)
{
stream->isPaused=FALSE;
#ifdef USE_GST1
GstSegment segment;
gst_segment_init(&segment, GST_FORMAT_TIME);
segment.start = pts;
segment.position = 0;
segment.rate = AAMP_NORMAL_PLAY_RATE;
segment.applied_rate = rate;
GST_INFO_OBJECT(aamp, "Sending segment event. start %" G_GUINT64_FORMAT " stop %" G_GUINT64_FORMAT" rate %f\n", segment.start, segment.stop, segment.rate);
GstEvent* event = gst_event_new_segment (&segment);
#else
GstEvent* event = gst_event_new_new_segment(FALSE, AAMP_NORMAL_PLAY_RATE, GST_FORMAT_TIME, pts, GST_CLOCK_TIME_NONE, 0);
#endif
gst_aamp_stream_add_item(stream, event);
stream->resetPosition = FALSE;
}
stream->eventsPending = FALSE;
}
/**
* @brief Sends stream data to src pad
* @param[in] mediaType MediaType of data
* @param[in] ptr Data buffer
* @param[in] len0 Size of data buffer
* @param[in] fpts PTS of buffer in seconds
* @param[in] fdts DTS of buffer in seconds
* @param[in] fDuration Duration of buffer in seconds
* @note Caller owns ptr, may free on return
*/
void Send(MediaType mediaType, const void *ptr, size_t len0, double fpts, double fdts, double fDuration)
{
GST_TRACE_OBJECT(aamp, "Enter Send mediaType %d, len0 %d, fpts %lf, fdts %lf, fDuration %lf\n", mediaType, len0, fpts, fdts, fDuration);
gboolean discontinuity = FALSE;
#ifdef AAMP_DISCARD_AUDIO_TRACK
if (mediaType == eMEDIATYPE_AUDIO)
{
GST_WARNING_OBJECT(aamp, "Discard audio track- not sending data\n");
return;
}
#endif
const char* mediaTypeStr = (mediaType==eMEDIATYPE_AUDIO)?"eMEDIATYPE_AUDIO":"eMEDIATYPE_VIDEO";
GST_DEBUG_OBJECT(aamp, "Enter len = %d fpts %f mediaType %s", (int)len0, fpts, mediaTypeStr);
if (!readyToSend)
{
if (!gst_aamp_ready(aamp))
{
GST_WARNING_OBJECT(aamp, "Not ready to consume data type %s\n", mediaTypeStr);
return;
}
readyToSend = true;
}
media_stream* stream = &aamp->stream[mediaType];
if (stream->isPaused)
{
for (int i = 0; i < STREAM_COUNT; i++)
aamp->stream[i].isPaused = TRUE;
}
if (stream->resetPosition && aamp->player_aamp->aamp->seek_pos_seconds > 0)
{
aamp->spts = aamp->player_aamp->aamp->seek_pos_seconds;
GST_DEBUG_OBJECT(aamp, "Updating spts %f mediaType %s", aamp->spts, mediaTypeStr);
}
if (aamp->spts > 0)
{
fpts += aamp->spts;
}
GST_TRACE_OBJECT(aamp, "Enter Send mediaType %d, Updated fpts %lf\n", mediaType, fpts);
GstClockTime pts = (GstClockTime)(fpts * GST_SECOND);
GstClockTime dts = (GstClockTime)(fdts * GST_SECOND);
//#define TRACE_PTS_TRACK 0xff
#ifdef TRACE_PTS_TRACK
if (( mediaType == TRACE_PTS_TRACK ) || (0xFF == TRACE_PTS_TRACK))
{
printf("%s : fpts %f pts %llu", (mediaType == eMEDIATYPE_VIDEO)?"vid":"aud", fpts, (unsigned long long)pts);
GstClock *clock = GST_ELEMENT_CLOCK(aamp);
if (clock)
{
GstClockTime curr = gst_clock_get_time(clock);
printf(" provided clock time %lu diff (pts - curr) %lu (%lu ms)", (unsigned long)curr, (unsigned long)(pts-curr), (unsigned long)(pts-curr)/GST_MSECOND);
}
printf("\n");
}
#endif
if(stream->eventsPending)
{
SendPendingEvents(stream , pts);
discontinuity = TRUE;
}
#ifdef GSTAAMP_DUMP_STREAM
static FILE* fp[2] = {NULL,NULL};
static char filename[128];
if (!fp[mediaType])
{
sprintf(filename, "gstaampdump%d.ts",mediaType );
fp[mediaType] = fopen(filename, "w");
}
fwrite(ptr, 1, len0, fp[mediaType] );
#endif
while (aamp->player_aamp->aamp->DownloadsAreEnabled() && !stream->isPaused && stream->srcpad)
{
size_t len = len0;
if (len > MAX_BYTES_TO_SEND)
{
len = MAX_BYTES_TO_SEND;
}
#ifdef USE_GST1
GstBuffer *buffer = gst_buffer_new_allocate(NULL, (gsize) len, NULL);
GstMapInfo map;
gst_buffer_map(buffer, &map, GST_MAP_WRITE);
memcpy(map.data, ptr, len);
gst_buffer_unmap(buffer, &map);
GST_BUFFER_PTS(buffer) = pts;
GST_BUFFER_DTS(buffer) = dts;
#else
GstBuffer *buffer = gst_buffer_new_and_alloc((guint) len);
memcpy(GST_BUFFER_DATA(buffer), ptr, len);
GST_BUFFER_TIMESTAMP(buffer) = pts;
#endif
if (discontinuity)
{
GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_FLAG_DISCONT);
discontinuity = FALSE;
}
gst_aamp_stream_add_item( stream, buffer);
ptr = len + (unsigned char *) ptr;
len0 -= len;
if (len0 == 0)
{
break;
}
}
GST_TRACE_OBJECT(aamp, "Exit");
}
/**
* @brief Sends stream data to src pad
* @param[in] mediaType MediaType of data
* @param[in] pBuffer Data buffer
* @param[in] fpts PTS of buffer in seconds
* @param[in] fdts DTS of buffer in seconds
* @param[in] fDuration Duration of buffer in seconds
* @note Ownership of pBuffer is transferred
*/
void Send(MediaType mediaType, GrowableBuffer* pBuffer, double fpts, double fdts, double fDuration)
{
GST_DEBUG_OBJECT(aamp, "Enter Send mediaType %d, fpts %lf, fdts %lf, fDuration %lf\n", mediaType, fpts, fdts, fDuration);
gboolean discontinuity = FALSE;
#ifdef AAMP_DISCARD_AUDIO_TRACK
if (mediaType == eMEDIATYPE_AUDIO)
{
GST_WARNING_OBJECT(aamp, "Discard audio track- not sending data\n");
return;
}
#endif
const char* mediaTypeStr = (mediaType == eMEDIATYPE_AUDIO) ? "eMEDIATYPE_AUDIO" : "eMEDIATYPE_VIDEO";
GST_INFO_OBJECT(aamp, "Enter len = %d fpts %f mediaType %s", (int) pBuffer->len, fpts, mediaTypeStr);
if (!readyToSend)
{
if (!gst_aamp_ready(aamp))
{
GST_WARNING_OBJECT(aamp, "Not ready to consume data type %s\n", mediaTypeStr);
return;
}
readyToSend = true;
}
media_stream* stream = &aamp->stream[mediaType];
if (!stream->srcpad)
{
GST_WARNING_OBJECT(aamp, "Pad NULL mediaType: %s (%d) len = %d fpts %f\n", mediaTypeStr, mediaType,
(int) pBuffer->len, fpts);
return;
}
GstClockTime pts = (GstClockTime)(fpts * GST_SECOND);
GstClockTime dts = (GstClockTime)(fdts * GST_SECOND);
//#define TRACE_PTS_TRACK 0xff
#ifdef TRACE_PTS_TRACK
if (( mediaType == TRACE_PTS_TRACK ) || (0xFF == TRACE_PTS_TRACK))
{
printf("%s : fpts %f pts %llu", (mediaType == eMEDIATYPE_VIDEO)?"vid":"aud", fpts, (unsigned long long)pts);
GstClock *clock = GST_ELEMENT_CLOCK(aamp);
if (clock)
{
GstClockTime curr = gst_clock_get_time(clock);
printf(" provided clock time %lu diff (pts - curr) %lu (%lu ms)", (unsigned long)curr, (unsigned long)(pts-curr), (unsigned long)(pts-curr)/GST_MSECOND);
}
printf("\n");
}
#endif
if (stream->eventsPending)
{
SendPendingEvents(stream, pts);
discontinuity = TRUE;
}
#ifdef GSTAAMP_DUMP_STREAM
static FILE* fp[2] =
{ NULL,NULL};
static char filename[128];
if (!fp[mediaType])
{
sprintf(filename, "gstaampdump%d.ts",mediaType );
fp[mediaType] = fopen(filename, "w");
}
fwrite(buffer->ptr, 1, buffer->len, fp[mediaType] );
#endif
if (aamp->player_aamp->aamp->DownloadsAreEnabled())
{
#ifdef USE_GST1
GstBuffer* buffer = gst_buffer_new_wrapped (pBuffer->ptr ,pBuffer->len);
GST_BUFFER_PTS(buffer) = pts;
GST_BUFFER_DTS(buffer) = dts;
#else
GstBuffer* buffer = gst_buffer_new();
GST_BUFFER_SIZE(buffer) = pBuffer->len;
GST_BUFFER_MALLOCDATA(buffer) = pBuffer->ptr;
GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA(buffer)
GST_BUFFER_TIMESTAMP(buffer) = pts;
GST_BUFFER_DURATION(buffer) = duration;
#endif
if (discontinuity)
{
GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_FLAG_DISCONT);
discontinuity = FALSE;
}
gst_aamp_stream_add_item( stream, buffer);
}
/*Since ownership of buffer is given to gstreamer, reset pBuffer */
memset(pBuffer, 0x00, sizeof(GrowableBuffer));
GST_TRACE_OBJECT(aamp, "Exit");
}
/**
* @brief Updates internal rate
* @param[in] rate Rate at which media is played back
*/
void UpdateRate(gdouble rate)
{
GST_INFO_OBJECT(aamp, "Enter UpdateRate rate = %lf", rate);
if ( rate != this->rate)
{
this->rate = rate;
}
}
/**
* @brief Handles stream specific EOS notification from aamp core
* @param[in] type Media type of the stream
*/
void EndOfStreamReached(MediaType type)
{
GST_WARNING_OBJECT(aamp, "MediaType %d", (int)type);
media_stream* stream = &aamp->stream[type];
if (stream->srcpad)
{
gst_aamp_stream_add_item( stream, gst_event_new_eos());
}
}
/**
* @brief Handles Discontinuity of stream
* @param[in] mediaType type of stream
* @retval always false
*/
bool Discontinuity(MediaType mediaType)
{
GST_INFO_OBJECT(aamp, "Enter Discontinuity, mediaType = %d", mediaType);
aamp->stream[mediaType].resetPosition = TRUE;
aamp->stream[mediaType].eventsPending = TRUE;
return false;
}
/**
* @brief Flush gstaamp streams
* @param[in] position seek position
* @param[in] rate playback rate
*/
void Flush(double position, int rate, bool shouldTearDown)
{
if (!aamp->seekFlush)
{
GST_INFO_OBJECT(aamp, "Enter Stream Flush position = %lf rate = %d shouldTearDown %d", position, rate, shouldTearDown);
for (int i = 0; i < STREAM_COUNT; i++)
{
aamp->stream[i].flush = TRUE;
aamp->stream[i].eventsPending = TRUE;
}
aamp->seekFlush = TRUE;
}
}
void Stop(bool keepLastFrame)
{
GST_INFO_OBJECT(aamp, "Enter Stream stop keepLastFrame %d", keepLastFrame);
for (int i = 0; i < STREAM_COUNT; i++)
{
aamp->stream[i].resetPosition = TRUE;
aamp->stream[i].flush = TRUE;
aamp->stream[i].eventsPending = TRUE;
}
aamp->seekFlush = TRUE;
}
unsigned long getCCDecoderHandle(void)
{
GST_DEBUG_OBJECT(aamp, "Enter getCCDecoderHandle");
GstStructure *structure;
GstQuery *query;
const GValue *val;
gboolean ret;
gpointer decoder_handle = NULL;
if (!aamp->stream[eMEDIATYPE_VIDEO].srcpad)
{
GST_DEBUG_OBJECT(aamp, "No video src pad available for AAMP plugin to query decoder handle\n");
}
else
{
structure = gst_structure_new("get_video_handle", "video_handle", G_TYPE_POINTER, 0, NULL);
#ifdef USE_GST1
query = gst_query_new_custom(GST_QUERY_CUSTOM, structure);
#else
query = gst_query_new_application(GST_QUERY_CUSTOM, structure);
#endif
ret = gst_pad_peer_query(aamp->stream[eMEDIATYPE_VIDEO].srcpad, query);
if (ret)
{
GST_DEBUG_OBJECT(aamp, "Video decoder handle queried successfully\n");
structure = (GstStructure *) gst_query_get_structure(query);
val = gst_structure_get_value(structure, "video_handle");
if (val == NULL)
{
GST_ERROR_OBJECT(aamp, "Unable to retrieve video decoder handle from structure\n");
}
else
{
decoder_handle = g_value_get_pointer(val);
GST_DEBUG_OBJECT(aamp, "video decoder handle: %x\n", decoder_handle);
}
}
else
{
GST_ERROR_OBJECT(aamp, "Video decoder handle query failed \n");
}
gst_query_unref(query);
}
return (unsigned long)decoder_handle;
}
void Stream(void)
{
GST_DEBUG_OBJECT(aamp, "Enter Stream()");
for (int i = 0; i < STREAM_COUNT; i++)
{
aamp->stream[i].resetPosition = TRUE;
aamp->stream[i].eventsPending = TRUE;
}
}
void SeekStreamSink(double position, double rate)
{
GST_DEBUG_OBJECT(aamp, "Enter SeekStreamSink()");
}
void StopBuffering(bool forceStop)
{
GST_DEBUG_OBJECT(aamp, "Enter StopBuffering()");
}
void Event(const AAMPEventPtr& event);
private:
GstAamp * aamp;
gdouble rate;
bool srcPadCapsSent;
StreamOutputFormat format;
StreamOutputFormat audioFormat;
bool readyToSend;
};
#define AAMP_TYPE_INIT_CODE { \
GST_DEBUG_CATEGORY_INIT (gst_aamp_debug_category, "aamp", 0, \
"debug category for aamp element"); \
}
static GstStaticPadTemplate gst_aamp_sink_template_hls = GST_STATIC_PAD_TEMPLATE("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS("application/x-hls;"
"application/x-aamp;"));
static GstStaticPadTemplate gst_aamp_sink_template = GST_STATIC_PAD_TEMPLATE("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS("application/x-aamp;"));
#define AAMP_SRC_CAPS_STR "video/mpegts, " \
" systemstream=(boolean)true, "\
" packetsize=(int)188;"
#define AAMP_SRC_AUDIO_CAPS_STR \
"audio/mpeg, " \
"mpegversion = (int) 1;" \
"audio/mpeg, " \
"mpegversion = (int) 2, " \
"stream-format = (string) adts; " \
"audio/x-ac3; audio/x-eac3;"
static GstStaticPadTemplate gst_aamp_src_template_video =
GST_STATIC_PAD_TEMPLATE ("video_%02x",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS(AAMP_SRC_CAPS_STR));
static GstStaticPadTemplate gst_aamp_src_template_audio =
GST_STATIC_PAD_TEMPLATE ("audio_%02x",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS(AAMP_SRC_AUDIO_CAPS_STR));
/* class initialization */
G_DEFINE_TYPE_WITH_CODE (GstAamp, gst_aamp, GST_TYPE_ELEMENT, AAMP_TYPE_INIT_CODE);
/**
* @brief Idle task to report decoder handle
* @param[in] user_data gstaamp pointer
* @retval G_SOURCE_REMOVE if handle received, G_SOURCE_CONTINUE otherwise
*/
static gboolean gst_report_video_decode_handle(gpointer user_data)
{
GstAamp *aamp = (GstAamp *) user_data;
GST_DEBUG_OBJECT(aamp, "Enter gst_report_video_decode_handle");
aamp->player_aamp->aamp->NotifyFirstFrameReceived();
aamp->decoder_idle_id = 0;
return G_SOURCE_REMOVE;
}
/**
* @brief Updates audio src pad states.
* @param[in] aamp gstaamp pointer
*/
static void gst_aamp_update_audio_src_pad(GstAamp * aamp)
{
GST_INFO_OBJECT(aamp, "Enter gst_aamp_update_audio_src_pad");
#ifndef AAMP_DISCARD_AUDIO_TRACK
if (NULL != aamp->stream[eMEDIATYPE_AUDIO].srcpad)
{
gboolean enable_audio;
if ( aamp->rate != AAMP_NORMAL_PLAY_RATE)
{
enable_audio = FALSE;
}
else
{
enable_audio = TRUE;
}
if (enable_audio && !aamp->audio_enabled)
{
GST_INFO_OBJECT(aamp, "Enable aud and add pad");
if (FALSE == gst_pad_set_active (aamp->stream[eMEDIATYPE_AUDIO].srcpad, TRUE))
{
GST_WARNING_OBJECT(aamp, "gst_pad_set_active failed");
}
if (FALSE == gst_element_add_pad(GST_ELEMENT(aamp), aamp->stream[eMEDIATYPE_AUDIO].srcpad))
{
GST_WARNING_OBJECT(aamp, "gst_element_add_pad stream[eMEDIATYPE_AUDIO].srcpad failed");
}
if (aamp->enable_src_tasks)
{
gst_aamp_stream_start(&aamp->stream[eMEDIATYPE_AUDIO]);
}
aamp->stream[eMEDIATYPE_AUDIO].streamStart = TRUE;
aamp->stream[eMEDIATYPE_AUDIO].eventsPending = TRUE;
aamp->audio_enabled = TRUE;
}
else if (!enable_audio && aamp->audio_enabled)
{
GST_INFO_OBJECT(aamp, "Disable aud and remove pad");
if (FALSE == gst_pad_set_active (aamp->stream[eMEDIATYPE_AUDIO].srcpad, FALSE))
{
GST_WARNING_OBJECT(aamp, "gst_pad_set_active FALSE failed");
}
if (FALSE == gst_element_remove_pad(GST_ELEMENT(aamp), aamp->stream[eMEDIATYPE_AUDIO].srcpad))
{
GST_WARNING_OBJECT(aamp, "gst_element_remove_pad stream[eMEDIATYPE_AUDIO].srcpad failed");
}
aamp->audio_enabled = FALSE;
}
}
#endif
}
/**
* @brief Creates GstCaps corresponding to stream format
* @param[in] format Output format of stream
* @retval GstCaps corresponding to stream format
* @note Caller shall free returned caps
*/
static GstCaps* GetGstCaps(StreamOutputFormat format)
{
GstCaps * caps = NULL;
switch (format)
{
case FORMAT_MPEGTS:
caps = gst_caps_new_simple ("video/mpegts",
"systemstream", G_TYPE_BOOLEAN, TRUE,
"packetsize", G_TYPE_INT, 188, NULL);
break;
case FORMAT_ISO_BMFF:
caps = gst_caps_new_simple("video/quicktime", NULL, NULL);
break;
case FORMAT_AUDIO_ES_AAC:
caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 2,
"stream-format", G_TYPE_STRING, "adts", NULL);
break;
case FORMAT_AUDIO_ES_AC3:
caps = gst_caps_new_simple ("audio/ac3", NULL, NULL);
break;
case FORMAT_AUDIO_ES_EC3:
caps = gst_caps_new_simple ("audio/x-eac3", NULL, NULL);
break;
case FORMAT_VIDEO_ES_H264:
caps = gst_caps_new_simple ("video/x-h264", NULL, NULL);
break;
case FORMAT_VIDEO_ES_MPEG2:
caps = gst_caps_new_simple ("video/mpeg",
"mpegversion", G_TYPE_INT, 2,
"systemstream", G_TYPE_BOOLEAN, FALSE, NULL);
break;
case FORMAT_UNKNOWN:
g_warning("Unknown format %d\n", format);
break;
case FORMAT_INVALID:
default:
g_warning("Unsupported format %d\n", format);
break;
}
return caps;
}
/**
* @brief Initialize a stream.
* @param[in] parent pointer to gstaamp instance
* @param[out] stream pointer to stream structure
*/
void gst_aamp_initialize_stream( GstAamp* parent, media_stream* stream)
{
GST_DEBUG_OBJECT(parent, "Enter gst_aamp_initialize_stream");
stream->queue = g_queue_new ();
stream->parent = parent;
g_mutex_init (&stream->mutex);
g_cond_init (&stream->cond);
}
/**
* @brief Configures gstaamp with stream output formats
* @param[in] aamp gstaamp pointer
* @param[in] format Output format of main media
* @param[in] audioFormat Output format of audio if present
*/
static void gst_aamp_configure(GstAamp * aamp, StreamOutputFormat format, StreamOutputFormat audioFormat)
{
GST_DEBUG_OBJECT(aamp, "Enter gst_aamp_configure format %d, audioFormat %d", format, audioFormat);
GstCaps *caps;
gchar * padname = NULL;
GST_DEBUG_OBJECT(aamp, "format %d audioFormat %d", format, audioFormat);
g_mutex_lock (&aamp->mutex);
if ( aamp->state >= GST_AAMP_CONFIGURED )
{
gst_aamp_update_audio_src_pad(aamp);
g_mutex_unlock (&aamp->mutex);
GST_INFO_OBJECT(aamp, "Already configured, sending streamStart");
return;
}
g_mutex_unlock (&aamp->mutex);
if (aamp->player_aamp->aamp->IsMuxedStream())
{
GST_INFO_OBJECT(aamp, "Muxed stream, enable src pad tasks");
aamp->enable_src_tasks = TRUE;
}
else
{
GST_INFO_OBJECT(aamp, "de-muxed stream, do not enable src pad tasks");
aamp->enable_src_tasks = FALSE;
}
caps = GetGstCaps(format);
if (caps)
{
media_stream* video = &aamp->stream[eMEDIATYPE_VIDEO];
padname = g_strdup_printf ("video_%02x", 1);
GstPad *srcpad = gst_pad_new_from_static_template(&gst_aamp_src_template_video, padname);
gst_object_ref(srcpad);
gst_pad_use_fixed_caps(srcpad);
GST_OBJECT_FLAG_SET(srcpad, GST_PAD_FLAG_NEED_PARENT);
gst_pad_set_query_function(srcpad, GST_DEBUG_FUNCPTR(gst_aamp_src_query));
gst_pad_set_event_function(srcpad, GST_DEBUG_FUNCPTR(gst_aamp_src_event));
video->caps = caps;
video->srcpad = srcpad;
gst_aamp_initialize_stream(aamp, video);
if (padname)
{
GST_INFO_OBJECT(aamp, "Created pad %s", padname);
g_free (padname);
}