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|
/*
* Copyright (C) 2007, 2009 Apple Inc. All rights reserved.
* Copyright (C) 2007 Collabora Ltd. All rights reserved.
* Copyright (C) 2007 Alp Toker <alp@atoker.com>
* Copyright (C) 2009 Gustavo Noronha Silva <gns@gnome.org>
* Copyright (C) 2009, 2010 Igalia S.L
*
* 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; either
* version 2 of the License, or (at your option) any later version.
*
* 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
* aint with this library; see the file COPYING.LIB. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include "config.h"
#if ENABLE(VIDEO)
#include "MediaPlayerPrivateGStreamer.h"
#include "ColorSpace.h"
#include "DataSourceGStreamer.h"
#include "Document.h"
#include "Frame.h"
#include "FrameView.h"
#include "GOwnPtrGStreamer.h"
#include "GStreamerGWorld.h"
#include "GraphicsContext.h"
#include "GraphicsTypes.h"
#include "ImageGStreamer.h"
#include "IntRect.h"
#include "KURL.h"
#include "MIMETypeRegistry.h"
#include "MediaPlayer.h"
#include "NotImplemented.h"
#include "ScrollView.h"
#include "SecurityOrigin.h"
#include "TimeRanges.h"
#include "VideoSinkGStreamer.h"
#include "WebKitWebSourceGStreamer.h"
#include "Widget.h"
#include <wtf/text/CString.h>
#include <GOwnPtr.h>
#include <gst/gst.h>
#include <gst/interfaces/mixer.h>
#include <gst/video/video.h>
#include <limits>
#include <math.h>
// GstPlayFlags flags from playbin2. It is the policy of GStreamer to
// not publicly expose element-specific enums. That's why this
// GstPlayFlags enum has been copied here.
typedef enum {
GST_PLAY_FLAG_VIDEO = 0x00000001,
GST_PLAY_FLAG_AUDIO = 0x00000002,
GST_PLAY_FLAG_TEXT = 0x00000004,
GST_PLAY_FLAG_VIS = 0x00000008,
GST_PLAY_FLAG_SOFT_VOLUME = 0x00000010,
GST_PLAY_FLAG_NATIVE_AUDIO = 0x00000020,
GST_PLAY_FLAG_NATIVE_VIDEO = 0x00000040,
GST_PLAY_FLAG_DOWNLOAD = 0x00000080,
GST_PLAY_FLAG_BUFFERING = 0x000000100
} GstPlayFlags;
using namespace std;
namespace WebCore {
static int greatestCommonDivisor(int a, int b)
{
while (b) {
int temp = a;
a = b;
b = temp % b;
}
return ABS(a);
}
gboolean mediaPlayerPrivateMessageCallback(GstBus* bus, GstMessage* message, gpointer data)
{
GOwnPtr<GError> err;
GOwnPtr<gchar> debug;
MediaPlayer::NetworkState error;
MediaPlayerPrivateGStreamer* mp = reinterpret_cast<MediaPlayerPrivateGStreamer*>(data);
bool issueError = true;
bool attemptNextLocation = false;
GstElement* pipeline = mp->pipeline();
if (message->structure) {
const gchar* messageTypeName = gst_structure_get_name(message->structure);
// Redirect messages are sent from elements, like qtdemux, to
// notify of the new location(s) of the media.
if (!g_strcmp0(messageTypeName, "redirect")) {
mp->mediaLocationChanged(message);
return true;
}
}
switch (GST_MESSAGE_TYPE(message)) {
case GST_MESSAGE_ERROR:
if (mp && mp->pipelineReset())
break;
gst_message_parse_error(message, &err.outPtr(), &debug.outPtr());
LOG_VERBOSE(Media, "Error: %d, %s", err->code, err->message);
error = MediaPlayer::Empty;
if (err->code == GST_STREAM_ERROR_CODEC_NOT_FOUND
|| err->code == GST_STREAM_ERROR_WRONG_TYPE
|| err->code == GST_STREAM_ERROR_FAILED
|| err->code == GST_CORE_ERROR_MISSING_PLUGIN
|| err->code == GST_RESOURCE_ERROR_NOT_FOUND)
error = MediaPlayer::FormatError;
else if (err->domain == GST_STREAM_ERROR) {
error = MediaPlayer::DecodeError;
attemptNextLocation = true;
} else if (err->domain == GST_RESOURCE_ERROR)
error = MediaPlayer::NetworkError;
if (mp) {
if (attemptNextLocation)
issueError = !mp->loadNextLocation();
if (issueError)
mp->loadingFailed(error);
}
break;
case GST_MESSAGE_EOS:
LOG_VERBOSE(Media, "End of Stream");
mp->didEnd();
break;
case GST_MESSAGE_STATE_CHANGED:
// Ignore state changes from internal elements. They are
// forwarded to playbin2 anyway.
if (GST_MESSAGE_SRC(message) == reinterpret_cast<GstObject*>(pipeline))
mp->updateStates();
break;
case GST_MESSAGE_BUFFERING:
mp->processBufferingStats(message);
break;
case GST_MESSAGE_DURATION:
LOG_VERBOSE(Media, "Duration changed");
mp->durationChanged();
break;
default:
LOG_VERBOSE(Media, "Unhandled GStreamer message type: %s",
GST_MESSAGE_TYPE_NAME(message));
break;
}
return true;
}
void mediaPlayerPrivateSourceChangedCallback(GObject *object, GParamSpec *pspec, gpointer data)
{
MediaPlayerPrivateGStreamer* mp = reinterpret_cast<MediaPlayerPrivateGStreamer*>(data);
GOwnPtr<GstElement> element;
g_object_get(mp->m_playBin, "source", &element.outPtr(), NULL);
gst_object_replace((GstObject**) &mp->m_source, (GstObject*) element.get());
if (WEBKIT_IS_WEB_SRC(element.get())) {
Frame* frame = mp->m_player->frameView() ? mp->m_player->frameView()->frame() : 0;
if (frame)
webKitWebSrcSetFrame(WEBKIT_WEB_SRC(element.get()), frame);
}
}
void mediaPlayerPrivateVolumeChangedCallback(GObject *element, GParamSpec *pspec, gpointer data)
{
// This is called when playbin receives the notify::volume signal.
MediaPlayerPrivateGStreamer* mp = reinterpret_cast<MediaPlayerPrivateGStreamer*>(data);
mp->volumeChanged();
}
void mediaPlayerPrivateMuteChangedCallback(GObject *element, GParamSpec *pspec, gpointer data)
{
// This is called when playbin receives the notify::mute signal.
MediaPlayerPrivateGStreamer* mp = reinterpret_cast<MediaPlayerPrivateGStreamer*>(data);
mp->muteChanged();
}
static float playbackPosition(GstElement* playbin)
{
float ret = 0.0f;
GstQuery* query = gst_query_new_position(GST_FORMAT_TIME);
if (!gst_element_query(playbin, query)) {
LOG_VERBOSE(Media, "Position query failed...");
gst_query_unref(query);
return ret;
}
gint64 position;
gst_query_parse_position(query, 0, &position);
// Position is available only if the pipeline is not in GST_STATE_NULL or
// GST_STATE_READY state.
if (position != static_cast<gint64>(GST_CLOCK_TIME_NONE))
ret = static_cast<float>(position) / static_cast<float>(GST_SECOND);
LOG_VERBOSE(Media, "Position %" GST_TIME_FORMAT, GST_TIME_ARGS(position));
gst_query_unref(query);
return ret;
}
void mediaPlayerPrivateRepaintCallback(WebKitVideoSink*, GstBuffer *buffer, MediaPlayerPrivateGStreamer* playerPrivate)
{
g_return_if_fail(GST_IS_BUFFER(buffer));
gst_buffer_replace(&playerPrivate->m_buffer, buffer);
playerPrivate->repaint();
}
MediaPlayerPrivateInterface* MediaPlayerPrivateGStreamer::create(MediaPlayer* player)
{
return new MediaPlayerPrivateGStreamer(player);
}
void MediaPlayerPrivateGStreamer::registerMediaEngine(MediaEngineRegistrar registrar)
{
if (isAvailable())
registrar(create, getSupportedTypes, supportsType);
}
static bool gstInitialized = false;
static bool doGstInit()
{
// FIXME: We should pass the arguments from the command line
if (!gstInitialized) {
GOwnPtr<GError> error;
gstInitialized = gst_init_check(0, 0, &error.outPtr());
if (!gstInitialized) {
LOG_VERBOSE(Media, "Could not initialize GStreamer: %s",
error ? error->message : "unknown error occurred");
} else {
gst_element_register(0, "webkitmediasrc", GST_RANK_PRIMARY,
WEBKIT_TYPE_DATA_SRC);
gst_element_register(0, "webkitwebsrc", GST_RANK_PRIMARY + 100,
WEBKIT_TYPE_WEB_SRC);
}
}
return gstInitialized;
}
bool MediaPlayerPrivateGStreamer::isAvailable()
{
if (!doGstInit())
return false;
GstElementFactory* factory = gst_element_factory_find("playbin2");
if (factory) {
gst_object_unref(GST_OBJECT(factory));
return true;
}
return false;
}
MediaPlayerPrivateGStreamer::MediaPlayerPrivateGStreamer(MediaPlayer* player)
: m_player(player)
, m_playBin(0)
, m_webkitVideoSink(0)
, m_fpsSink(0)
, m_source(0)
, m_seekTime(0)
, m_changingRate(false)
, m_endTime(numeric_limits<float>::infinity())
, m_networkState(MediaPlayer::Empty)
, m_readyState(MediaPlayer::HaveNothing)
, m_isStreaming(false)
, m_size(IntSize())
, m_buffer(0)
, m_mediaLocations(0)
, m_mediaLocationCurrentIndex(0)
, m_resetPipeline(false)
, m_paused(true)
, m_seeking(false)
, m_buffering(false)
, m_playbackRate(1)
, m_errorOccured(false)
, m_mediaDuration(0)
, m_startedBuffering(false)
, m_fillTimer(this, &MediaPlayerPrivateGStreamer::fillTimerFired)
, m_maxTimeLoaded(0)
, m_bufferingPercentage(0)
, m_preload(MediaPlayer::Auto)
, m_delayingLoad(false)
, m_mediaDurationKnown(true)
{
if (doGstInit())
createGSTPlayBin();
}
MediaPlayerPrivateGStreamer::~MediaPlayerPrivateGStreamer()
{
if (m_fillTimer.isActive())
m_fillTimer.stop();
if (m_buffer)
gst_buffer_unref(m_buffer);
m_buffer = 0;
if (m_mediaLocations) {
gst_structure_free(m_mediaLocations);
m_mediaLocations = 0;
}
if (m_source) {
gst_object_unref(m_source);
m_source = 0;
}
if (m_videoSinkBin) {
gst_object_unref(m_videoSinkBin);
m_videoSinkBin = 0;
}
if (m_playBin) {
gst_element_set_state(m_playBin, GST_STATE_NULL);
gst_object_unref(GST_OBJECT(m_playBin));
}
}
void MediaPlayerPrivateGStreamer::load(const String& url)
{
g_object_set(m_playBin, "uri", url.utf8().data(), NULL);
LOG_VERBOSE(Media, "Load %s", url.utf8().data());
if (m_preload == MediaPlayer::None) {
LOG_VERBOSE(Media, "Delaying load.");
m_delayingLoad = true;
return;
}
commitLoad();
}
void MediaPlayerPrivateGStreamer::commitLoad()
{
// GStreamer needs to have the pipeline set to a paused state to
// start providing anything useful.
gst_element_set_state(m_playBin, GST_STATE_PAUSED);
LOG_VERBOSE(Media, "Committing load.");
if (m_networkState != MediaPlayer::Loading) {
m_networkState = MediaPlayer::Loading;
m_player->networkStateChanged();
}
if (m_readyState != MediaPlayer::HaveNothing) {
m_readyState = MediaPlayer::HaveNothing;
m_player->readyStateChanged();
}
}
bool MediaPlayerPrivateGStreamer::changePipelineState(GstState newState)
{
ASSERT(newState == GST_STATE_PLAYING || newState == GST_STATE_PAUSED);
GstState currentState;
GstState pending;
gst_element_get_state(m_playBin, ¤tState, &pending, 0);
if (currentState != newState && pending != newState) {
GstStateChangeReturn ret = gst_element_set_state(m_playBin, newState);
GstState pausedOrPlaying = newState == GST_STATE_PLAYING ? GST_STATE_PAUSED : GST_STATE_PLAYING;
if (currentState != pausedOrPlaying && ret == GST_STATE_CHANGE_FAILURE) {
loadingFailed(MediaPlayer::Empty);
return false;
}
}
return true;
}
void MediaPlayerPrivateGStreamer::prepareToPlay()
{
if (m_delayingLoad) {
m_delayingLoad = false;
commitLoad();
}
}
void MediaPlayerPrivateGStreamer::play()
{
if (changePipelineState(GST_STATE_PLAYING))
LOG_VERBOSE(Media, "Play");
}
void MediaPlayerPrivateGStreamer::pause()
{
if (changePipelineState(GST_STATE_PAUSED))
LOG_VERBOSE(Media, "Pause");
}
float MediaPlayerPrivateGStreamer::duration() const
{
if (!m_playBin)
return 0.0f;
if (m_errorOccured)
return 0.0f;
// Media duration query failed already, don't attempt new useless queries.
if (!m_mediaDurationKnown)
return numeric_limits<float>::infinity();
if (m_mediaDuration)
return m_mediaDuration;
GstFormat timeFormat = GST_FORMAT_TIME;
gint64 timeLength = 0;
if (!gst_element_query_duration(m_playBin, &timeFormat, &timeLength) || timeFormat != GST_FORMAT_TIME || static_cast<guint64>(timeLength) == GST_CLOCK_TIME_NONE) {
LOG_VERBOSE(Media, "Time duration query failed.");
return numeric_limits<float>::infinity();
}
LOG_VERBOSE(Media, "Duration: %" GST_TIME_FORMAT, GST_TIME_ARGS(timeLength));
return (float) ((guint64) timeLength / 1000000000.0);
// FIXME: handle 3.14.9.5 properly
}
float MediaPlayerPrivateGStreamer::currentTime() const
{
if (!m_playBin)
return 0.0f;
if (m_errorOccured)
return 0.0f;
if (m_seeking)
return static_cast<float>(m_seekTime);
return playbackPosition(m_playBin);
}
void MediaPlayerPrivateGStreamer::seek(float time)
{
// Avoid useless seeking.
if (time == playbackPosition(m_playBin))
return;
if (!m_playBin)
return;
if (m_errorOccured)
return;
GstClockTime sec = (GstClockTime)(static_cast<float>(time * GST_SECOND));
LOG_VERBOSE(Media, "Seek: %" GST_TIME_FORMAT, GST_TIME_ARGS(sec));
if (!gst_element_seek(m_playBin, m_player->rate(),
GST_FORMAT_TIME,
(GstSeekFlags)(GST_SEEK_FLAG_FLUSH),
GST_SEEK_TYPE_SET, sec,
GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
LOG_VERBOSE(Media, "Seek to %f failed", time);
else {
m_seeking = true;
m_seekTime = sec;
}
}
void MediaPlayerPrivateGStreamer::startEndPointTimerIfNeeded()
{
notImplemented();
}
void MediaPlayerPrivateGStreamer::cancelSeek()
{
notImplemented();
}
void MediaPlayerPrivateGStreamer::endPointTimerFired(Timer<MediaPlayerPrivateGStreamer>*)
{
notImplemented();
}
bool MediaPlayerPrivateGStreamer::paused() const
{
return m_paused;
}
bool MediaPlayerPrivateGStreamer::seeking() const
{
return m_seeking;
}
// Returns the size of the video
IntSize MediaPlayerPrivateGStreamer::naturalSize() const
{
if (!hasVideo())
return IntSize();
GstPad* pad = gst_element_get_static_pad(m_webkitVideoSink, "sink");
if (!pad)
return IntSize();
guint64 width = 0, height = 0;
GstCaps* caps = GST_PAD_CAPS(pad);
int pixelAspectRatioNumerator, pixelAspectRatioDenominator;
int displayWidth, displayHeight, displayAspectRatioGCD;
int originalWidth = 0, originalHeight = 0;
// TODO: handle possible clean aperture data. See
// https://bugzilla.gnome.org/show_bug.cgi?id=596571
// TODO: handle possible transformation matrix. See
// https://bugzilla.gnome.org/show_bug.cgi?id=596326
// Get the video PAR and original size.
if (!GST_IS_CAPS(caps) || !gst_caps_is_fixed(caps)
|| !gst_video_format_parse_caps(caps, 0, &originalWidth, &originalHeight)
|| !gst_video_parse_caps_pixel_aspect_ratio(caps, &pixelAspectRatioNumerator,
&pixelAspectRatioDenominator)) {
gst_object_unref(GST_OBJECT(pad));
return IntSize();
}
gst_object_unref(GST_OBJECT(pad));
LOG_VERBOSE(Media, "Original video size: %dx%d", originalWidth, originalHeight);
LOG_VERBOSE(Media, "Pixel aspect ratio: %d/%d", pixelAspectRatioNumerator, pixelAspectRatioDenominator);
// Calculate DAR based on PAR and video size.
displayWidth = originalWidth * pixelAspectRatioNumerator;
displayHeight = originalHeight * pixelAspectRatioDenominator;
// Divide display width and height by their GCD to avoid possible overflows.
displayAspectRatioGCD = greatestCommonDivisor(displayWidth, displayHeight);
displayWidth /= displayAspectRatioGCD;
displayHeight /= displayAspectRatioGCD;
// Apply DAR to original video size. This is the same behavior as in xvimagesink's setcaps function.
if (!(originalHeight % displayHeight)) {
LOG_VERBOSE(Media, "Keeping video original height");
width = gst_util_uint64_scale_int(originalHeight, displayWidth, displayHeight);
height = static_cast<guint64>(originalHeight);
} else if (!(originalWidth % displayWidth)) {
LOG_VERBOSE(Media, "Keeping video original width");
height = gst_util_uint64_scale_int(originalWidth, displayHeight, displayWidth);
width = static_cast<guint64>(originalWidth);
} else {
LOG_VERBOSE(Media, "Approximating while keeping original video height");
width = gst_util_uint64_scale_int(originalHeight, displayWidth, displayHeight);
height = static_cast<guint64>(originalHeight);
}
LOG_VERBOSE(Media, "Natural size: %" G_GUINT64_FORMAT "x%" G_GUINT64_FORMAT, width, height);
return IntSize(static_cast<int>(width), static_cast<int>(height));
}
bool MediaPlayerPrivateGStreamer::hasVideo() const
{
gint currentVideo = -1;
if (m_playBin)
g_object_get(m_playBin, "current-video", ¤tVideo, NULL);
return currentVideo > -1;
}
bool MediaPlayerPrivateGStreamer::hasAudio() const
{
gint currentAudio = -1;
if (m_playBin)
g_object_get(m_playBin, "current-audio", ¤tAudio, NULL);
return currentAudio > -1;
}
void MediaPlayerPrivateGStreamer::setVolume(float volume)
{
if (!m_playBin)
return;
g_object_set(m_playBin, "volume", static_cast<double>(volume), NULL);
}
void MediaPlayerPrivateGStreamer::volumeChangedTimerFired(Timer<MediaPlayerPrivateGStreamer>*)
{
double volume;
g_object_get(m_playBin, "volume", &volume, NULL);
m_player->volumeChanged(static_cast<float>(volume));
}
void MediaPlayerPrivateGStreamer::volumeChanged()
{
Timer<MediaPlayerPrivateGStreamer> volumeChangedTimer(this, &MediaPlayerPrivateGStreamer::volumeChangedTimerFired);
volumeChangedTimer.startOneShot(0);
}
void MediaPlayerPrivateGStreamer::setRate(float rate)
{
// Avoid useless playback rate update.
if (m_playbackRate == rate)
return;
GstState state;
GstState pending;
gst_element_get_state(m_playBin, &state, &pending, 0);
if ((state != GST_STATE_PLAYING && state != GST_STATE_PAUSED)
|| (pending == GST_STATE_PAUSED))
return;
if (m_isStreaming)
return;
m_playbackRate = rate;
m_changingRate = true;
float currentPosition = static_cast<float>(playbackPosition(m_playBin) * GST_SECOND);
GstSeekFlags flags = (GstSeekFlags)(GST_SEEK_FLAG_FLUSH);
gint64 start, end;
bool mute = false;
LOG_VERBOSE(Media, "Set Rate to %f", rate);
if (rate >= 0) {
// Mute the sound if the playback rate is too extreme.
// TODO: in other cases we should perform pitch adjustments.
mute = (bool) (rate < 0.8 || rate > 2);
start = currentPosition;
end = GST_CLOCK_TIME_NONE;
} else {
start = 0;
mute = true;
// If we are at beginning of media, start from the end to
// avoid immediate EOS.
if (currentPosition <= 0)
end = static_cast<gint64>(duration() * GST_SECOND);
else
end = currentPosition;
}
LOG_VERBOSE(Media, "Need to mute audio: %d", (int) mute);
if (!gst_element_seek(m_playBin, rate, GST_FORMAT_TIME, flags,
GST_SEEK_TYPE_SET, start,
GST_SEEK_TYPE_SET, end))
LOG_VERBOSE(Media, "Set rate to %f failed", rate);
else
g_object_set(m_playBin, "mute", mute, NULL);
}
MediaPlayer::NetworkState MediaPlayerPrivateGStreamer::networkState() const
{
return m_networkState;
}
MediaPlayer::ReadyState MediaPlayerPrivateGStreamer::readyState() const
{
return m_readyState;
}
PassRefPtr<TimeRanges> MediaPlayerPrivateGStreamer::buffered() const
{
RefPtr<TimeRanges> timeRanges = TimeRanges::create();
float loaded = maxTimeLoaded();
if (!m_errorOccured && !m_isStreaming && loaded > 0)
timeRanges->add(0, loaded);
return timeRanges.release();
}
void MediaPlayerPrivateGStreamer::processBufferingStats(GstMessage* message)
{
// This is the immediate buffering that needs to happen so we have
// enough to play right now.
m_buffering = true;
const GstStructure *structure = gst_message_get_structure(message);
gst_structure_get_int(structure, "buffer-percent", &m_bufferingPercentage);
LOG_VERBOSE(Media, "[Buffering] Buffering: %d%%.", m_bufferingPercentage);
GstBufferingMode mode;
gst_message_parse_buffering_stats(message, &mode, 0, 0, 0);
if (mode != GST_BUFFERING_DOWNLOAD) {
updateStates();
return;
}
// This is on-disk buffering, that allows us to download much more
// than needed for right now.
if (!m_startedBuffering) {
LOG_VERBOSE(Media, "[Buffering] Starting on-disk buffering.");
m_startedBuffering = true;
if (m_fillTimer.isActive())
m_fillTimer.stop();
m_fillTimer.startRepeating(0.2);
}
}
void MediaPlayerPrivateGStreamer::fillTimerFired(Timer<MediaPlayerPrivateGStreamer>*)
{
GstQuery* query = gst_query_new_buffering(GST_FORMAT_PERCENT);
if (!gst_element_query(m_playBin, query)) {
gst_query_unref(query);
return;
}
gint64 start, stop;
gdouble fillStatus = 100.0;
gst_query_parse_buffering_range(query, 0, &start, &stop, 0);
gst_query_unref(query);
if (stop != -1)
fillStatus = 100.0 * stop / GST_FORMAT_PERCENT_MAX;
LOG_VERBOSE(Media, "[Buffering] Download buffer filled up to %f%%", fillStatus);
if (!m_mediaDuration)
durationChanged();
// Update maxTimeLoaded only if the media duration is
// available. Otherwise we can't compute it.
if (m_mediaDuration) {
if (fillStatus == 100.0)
m_maxTimeLoaded = m_mediaDuration;
else
m_maxTimeLoaded = static_cast<float>((fillStatus * m_mediaDuration) / 100.0);
LOG_VERBOSE(Media, "[Buffering] Updated maxTimeLoaded: %f", m_maxTimeLoaded);
}
if (fillStatus != 100.0) {
updateStates();
return;
}
// Media is now fully loaded. It will play even if network
// connection is cut. Buffering is done, remove the fill source
// from the main loop.
m_fillTimer.stop();
m_startedBuffering = false;
updateStates();
}
float MediaPlayerPrivateGStreamer::maxTimeSeekable() const
{
if (m_errorOccured)
return 0.0f;
LOG_VERBOSE(Media, "maxTimeSeekable");
// infinite duration means live stream
if (isinf(duration()))
return 0.0f;
return duration();
}
float MediaPlayerPrivateGStreamer::maxTimeLoaded() const
{
if (m_errorOccured)
return 0.0f;
float loaded = m_maxTimeLoaded;
if (!loaded && !m_fillTimer.isActive())
loaded = duration();
LOG_VERBOSE(Media, "maxTimeLoaded: %f", loaded);
return loaded;
}
unsigned MediaPlayerPrivateGStreamer::bytesLoaded() const
{
if (!m_playBin)
return 0;
if (!m_mediaDuration)
return 0;
unsigned loaded = totalBytes() * maxTimeLoaded() / m_mediaDuration;
LOG_VERBOSE(Media, "bytesLoaded: %d", loaded);
return loaded;
}
unsigned MediaPlayerPrivateGStreamer::totalBytes() const
{
if (!m_source)
return 0;
if (m_errorOccured)
return 0;
GstFormat fmt = GST_FORMAT_BYTES;
gint64 length = 0;
gst_element_query_duration(m_source, &fmt, &length);
LOG_VERBOSE(Media, "totalBytes %" G_GINT64_FORMAT, length);
return static_cast<unsigned>(length);
}
void MediaPlayerPrivateGStreamer::cancelLoad()
{
if (m_networkState < MediaPlayer::Loading || m_networkState == MediaPlayer::Loaded)
return;
if (m_playBin)
gst_element_set_state(m_playBin, GST_STATE_NULL);
}
void MediaPlayerPrivateGStreamer::updateStates()
{
if (!m_playBin)
return;
if (m_errorOccured)
return;
MediaPlayer::NetworkState oldNetworkState = m_networkState;
MediaPlayer::ReadyState oldReadyState = m_readyState;
GstState state;
GstState pending;
GstStateChangeReturn ret = gst_element_get_state(m_playBin,
&state, &pending, 250 * GST_NSECOND);
bool shouldUpdateAfterSeek = false;
switch (ret) {
case GST_STATE_CHANGE_SUCCESS:
LOG_VERBOSE(Media, "State: %s, pending: %s",
gst_element_state_get_name(state),
gst_element_state_get_name(pending));
m_resetPipeline = state <= GST_STATE_READY;
// Try to figure out ready and network states.
if (state == GST_STATE_READY) {
m_readyState = MediaPlayer::HaveNothing;
m_networkState = MediaPlayer::Empty;
} else if (maxTimeLoaded() == duration()) {
m_networkState = MediaPlayer::Loaded;
m_readyState = MediaPlayer::HaveEnoughData;
} else {
m_readyState = currentTime() < maxTimeLoaded() ? MediaPlayer::HaveFutureData : MediaPlayer::HaveCurrentData;
m_networkState = MediaPlayer::Loading;
}
if (m_buffering && state != GST_STATE_READY) {
m_readyState = MediaPlayer::HaveCurrentData;
m_networkState = MediaPlayer::Loading;
}
// Now let's try to get the states in more detail using
// information from GStreamer, while we sync states where
// needed.
if (state == GST_STATE_PAUSED) {
if (m_buffering && m_bufferingPercentage == 100) {
m_buffering = false;
m_bufferingPercentage = 0;
m_readyState = MediaPlayer::HaveEnoughData;
LOG_VERBOSE(Media, "[Buffering] Complete.");
if (!m_paused) {
LOG_VERBOSE(Media, "[Buffering] Restarting playback.");
gst_element_set_state(m_playBin, GST_STATE_PLAYING);
}
} else if (!m_buffering && (currentTime() < duration())) {
m_paused = true;
}
} else if (state == GST_STATE_PLAYING) {
m_readyState = MediaPlayer::HaveEnoughData;
m_paused = false;
if (!m_mediaDuration) {
float newDuration = duration();
m_mediaDurationKnown = !isinf(newDuration);
if (m_mediaDurationKnown)
m_mediaDuration = newDuration;
}
if (m_buffering) {
m_readyState = MediaPlayer::HaveCurrentData;
m_networkState = MediaPlayer::Loading;
LOG_VERBOSE(Media, "[Buffering] Pausing stream for buffering.");
gst_element_set_state(m_playBin, GST_STATE_PAUSED);
}
} else
m_paused = true;
// Is on-disk buffering in progress?
if (m_fillTimer.isActive())
m_networkState = MediaPlayer::Loading;
if (m_changingRate) {
m_player->rateChanged();
m_changingRate = false;
}
if (m_seeking) {
shouldUpdateAfterSeek = true;
m_seeking = false;
}
break;
case GST_STATE_CHANGE_ASYNC:
LOG_VERBOSE(Media, "Async: State: %s, pending: %s",
gst_element_state_get_name(state),
gst_element_state_get_name(pending));
// Change in progress
if (!m_isStreaming && !m_buffering)
return;
// Resume playback if a seek was performed in a live pipeline
// or during progressive download. That second use-case
// happens when the seek is performed to a region of the media
// that hasn't been downloaded yet.
if (m_seeking) {
shouldUpdateAfterSeek = true;
m_seeking = false;
if (m_paused)
gst_element_set_state(m_playBin, GST_STATE_PLAYING);
}
break;
case GST_STATE_CHANGE_FAILURE:
LOG_VERBOSE(Media, "Failure: State: %s, pending: %s",
gst_element_state_get_name(state),
gst_element_state_get_name(pending));
// Change failed
return;
case GST_STATE_CHANGE_NO_PREROLL:
LOG_VERBOSE(Media, "No preroll: State: %s, pending: %s",
gst_element_state_get_name(state),
gst_element_state_get_name(pending));
if (state == GST_STATE_READY)
m_readyState = MediaPlayer::HaveNothing;
else if (state == GST_STATE_PAUSED) {
m_readyState = MediaPlayer::HaveEnoughData;
m_paused = true;
// Live pipelines go in PAUSED without prerolling.
m_isStreaming = true;
} else if (state == GST_STATE_PLAYING)
m_paused = false;
if (m_seeking) {
shouldUpdateAfterSeek = true;
m_seeking = false;
if (!m_paused)
gst_element_set_state(m_playBin, GST_STATE_PLAYING);
} else if (!m_paused)
gst_element_set_state(m_playBin, GST_STATE_PLAYING);
m_networkState = MediaPlayer::Loading;
break;
default:
LOG_VERBOSE(Media, "Else : %d", ret);
break;
}
if (seeking())
m_readyState = MediaPlayer::HaveNothing;
if (shouldUpdateAfterSeek)
timeChanged();
if (m_networkState != oldNetworkState) {
LOG_VERBOSE(Media, "Network State Changed from %u to %u",
oldNetworkState, m_networkState);
m_player->networkStateChanged();
}
if (m_readyState != oldReadyState) {
LOG_VERBOSE(Media, "Ready State Changed from %u to %u",
oldReadyState, m_readyState);
m_player->readyStateChanged();
}
}
void MediaPlayerPrivateGStreamer::mediaLocationChanged(GstMessage* message)
{
if (m_mediaLocations)
gst_structure_free(m_mediaLocations);
if (message->structure) {
// This structure can contain:
// - both a new-location string and embedded locations structure
// - or only a new-location string.
m_mediaLocations = gst_structure_copy(message->structure);
const GValue* locations = gst_structure_get_value(m_mediaLocations, "locations");
if (locations)
m_mediaLocationCurrentIndex = static_cast<int>(gst_value_list_get_size(locations)) -1;
loadNextLocation();
}
}
bool MediaPlayerPrivateGStreamer::loadNextLocation()
{
if (!m_mediaLocations)
return false;
const GValue* locations = gst_structure_get_value(m_mediaLocations, "locations");
const gchar* newLocation = 0;
if (!locations) {
// Fallback on new-location string.
newLocation = gst_structure_get_string(m_mediaLocations, "new-location");
if (!newLocation)
return false;
}
if (!newLocation) {
if (m_mediaLocationCurrentIndex < 0) {
m_mediaLocations = 0;
return false;
}
const GValue* location = gst_value_list_get_value(locations,
m_mediaLocationCurrentIndex);
const GstStructure* structure = gst_value_get_structure(location);
if (!structure) {
m_mediaLocationCurrentIndex--;
return false;
}
newLocation = gst_structure_get_string(structure, "new-location");
}
if (newLocation) {
// Found a candidate. new-location is not always an absolute url
// though. We need to take the base of the current url and
// append the value of new-location to it.
gchar* currentLocation = 0;
g_object_get(m_playBin, "uri", ¤tLocation, NULL);
KURL currentUrl(KURL(), currentLocation);
g_free(currentLocation);
KURL newUrl;
if (gst_uri_is_valid(newLocation))
newUrl = KURL(KURL(), newLocation);
else
newUrl = KURL(KURL(), currentUrl.baseAsString() + newLocation);
RefPtr<SecurityOrigin> securityOrigin = SecurityOrigin::create(currentUrl);
if (securityOrigin->canRequest(newUrl)) {
LOG_VERBOSE(Media, "New media url: %s", newUrl.string().utf8().data());
// Reset player states.
m_networkState = MediaPlayer::Loading;
m_player->networkStateChanged();
m_readyState = MediaPlayer::HaveNothing;
m_player->readyStateChanged();
// Reset pipeline state.
m_resetPipeline = true;
gst_element_set_state(m_playBin, GST_STATE_READY);
GstState state;
gst_element_get_state(m_playBin, &state, 0, 0);
if (state <= GST_STATE_READY) {
// Set the new uri and start playing.
g_object_set(m_playBin, "uri", newUrl.string().utf8().data(), NULL);
gst_element_set_state(m_playBin, GST_STATE_PLAYING);
return true;
}
}
}
m_mediaLocationCurrentIndex--;
return false;
}
void MediaPlayerPrivateGStreamer::loadStateChanged()
{
updateStates();
}
void MediaPlayerPrivateGStreamer::sizeChanged()
{
notImplemented();
}
void MediaPlayerPrivateGStreamer::timeChanged()
{
updateStates();
m_player->timeChanged();
}
void MediaPlayerPrivateGStreamer::didEnd()
{
// EOS was reached but in case of reverse playback the position is
// not always 0. So to not confuse the HTMLMediaElement we
// synchronize position and duration values.
float now = currentTime();
if (now > 0) {
m_mediaDuration = now;
m_mediaDurationKnown = true;
m_player->durationChanged();
}
gst_element_set_state(m_playBin, GST_STATE_PAUSED);
timeChanged();
}
void MediaPlayerPrivateGStreamer::durationChanged()
{
// Reset cached media duration
m_mediaDuration = 0;
// And re-cache it if possible.
GstState state;
gst_element_get_state(m_playBin, &state, 0, 0);
float newDuration = duration();
if (state <= GST_STATE_READY) {
// Don't set m_mediaDurationKnown yet if the pipeline is not
// paused. This allows duration() query to fail at least once
// before playback starts and duration becomes known.
if (!isinf(newDuration))
m_mediaDuration = newDuration;
} else {
m_mediaDurationKnown = !isinf(newDuration);
if (m_mediaDurationKnown)
m_mediaDuration = newDuration;
}
if (!isinf(newDuration))
m_mediaDuration = newDuration;
m_player->durationChanged();
}
bool MediaPlayerPrivateGStreamer::supportsMuting() const
{
return true;
}
void MediaPlayerPrivateGStreamer::setMuted(bool muted)
{
if (!m_playBin)
return;
g_object_set(m_playBin, "mute", muted, NULL);
}
void MediaPlayerPrivateGStreamer::muteChangedTimerFired(Timer<MediaPlayerPrivateGStreamer>*)
{
gboolean muted;
g_object_get(m_playBin, "mute", &muted, NULL);
m_player->muteChanged(static_cast<bool>(muted));
}
void MediaPlayerPrivateGStreamer::muteChanged()
{
Timer<MediaPlayerPrivateGStreamer> muteChangedTimer(this, &MediaPlayerPrivateGStreamer::muteChangedTimerFired);
muteChangedTimer.startOneShot(0);
}
void MediaPlayerPrivateGStreamer::loadingFailed(MediaPlayer::NetworkState error)
{
m_errorOccured = true;
if (m_networkState != error) {
m_networkState = error;
m_player->networkStateChanged();
}
if (m_readyState != MediaPlayer::HaveNothing) {
m_readyState = MediaPlayer::HaveNothing;
m_player->readyStateChanged();
}
}
void MediaPlayerPrivateGStreamer::setSize(const IntSize& size)
{
m_size = size;
}
void MediaPlayerPrivateGStreamer::setVisible(bool visible)
{
}
void MediaPlayerPrivateGStreamer::repaint()
{
m_player->repaint();
}
void MediaPlayerPrivateGStreamer::paint(GraphicsContext* context, const IntRect& rect)
{
if (context->paintingDisabled())
return;
if (!m_player->visible())
return;
if (!m_buffer)
return;
RefPtr<ImageGStreamer> gstImage = ImageGStreamer::createImage(m_buffer);
if (!gstImage)
return;
context->drawImage(reinterpret_cast<Image*>(gstImage->image().get()), sRGBColorSpace,
rect, CompositeCopy, false);
}
static HashSet<String> mimeTypeCache()
{
doGstInit();
DEFINE_STATIC_LOCAL(HashSet<String>, cache, ());
static bool typeListInitialized = false;
if (!typeListInitialized) {
// Build a whitelist of mime-types known to be supported by
// GStreamer.
HashSet<String> handledApplicationSubtypes;
handledApplicationSubtypes.add(String("ogg"));
handledApplicationSubtypes.add(String("x-3gp"));
handledApplicationSubtypes.add(String("vnd.rn-realmedia"));
handledApplicationSubtypes.add(String("x-pn-realaudio"));
GList* factories = gst_type_find_factory_get_list();
for (GList* iterator = factories; iterator; iterator = iterator->next) {
GstTypeFindFactory* factory = GST_TYPE_FIND_FACTORY(iterator->data);
GstCaps* caps = gst_type_find_factory_get_caps(factory);
if (!caps)
continue;
for (guint structureIndex = 0; structureIndex < gst_caps_get_size(caps); structureIndex++) {
GstStructure* structure = gst_caps_get_structure(caps, structureIndex);
const gchar* name = gst_structure_get_name(structure);
bool cached = false;
// These formats are supported by GStreamer, but not
// correctly advertised.
if (g_str_equal(name, "video/x-h264")
|| g_str_equal(name, "audio/x-m4a")) {
cache.add(String("video/mp4"));
cache.add(String("audio/aac"));
cache.add(String("audio/mp4"));
cached = true;
}
if (g_str_equal(name, "video/x-theora")) {
cache.add(String("video/ogg"));
cached = true;
}
if (g_str_equal(name, "audio/x-vorbis")) {
cache.add(String("audio/ogg"));
cached = true;
}
if (g_str_equal(name, "audio/x-wav")) {
cache.add(String("audio/wav"));
cache.add(String("audio/x-wav"));
cached = true;
}
if (g_str_equal(name, "audio/mpeg")) {
cache.add(String(name));
cache.add(String("audio/x-mpeg"));
cached = true;
// This is what we are handling:
// mpegversion=(int)1, layer=(int)[ 1, 3 ]
gint mpegVersion = 0;
if (gst_structure_get_int(structure, "mpegversion", &mpegVersion) && (mpegVersion == 1)) {
const GValue* layer = gst_structure_get_value(structure, "layer");
if (G_VALUE_TYPE(layer) == GST_TYPE_INT_RANGE) {
gint minLayer = gst_value_get_int_range_min(layer);
gint maxLayer = gst_value_get_int_range_max(layer);
if (minLayer <= 1 && 1 <= maxLayer)
cache.add(String("audio/mp1"));
if (minLayer <= 2 && 2 <= maxLayer)
cache.add(String("audio/mp2"));
if (minLayer <= 3 && 3 <= maxLayer) {
cache.add(String("audio/x-mp3"));
cache.add(String("audio/mp3"));
}
}
}
}
if (!cached) {
// GStreamer plugins can be capable of supporting
// types which WebKit supports by default. In that
// case, we should not consider these types
// supportable by GStreamer. Examples of what
// GStreamer can support but should not be added:
// text/plain, text/html, image/jpeg,
// application/xml
gchar** mimetype = g_strsplit(name, "/", 2);
if (g_str_equal(mimetype[0], "audio")
|| g_str_equal(mimetype[0], "video")
|| (g_str_equal(mimetype[0], "application")
&& handledApplicationSubtypes.contains(String(mimetype[1]))))
cache.add(String(name));
g_strfreev(mimetype);
}
}
}
gst_plugin_feature_list_free(factories);
typeListInitialized = true;
}
return cache;
}
void MediaPlayerPrivateGStreamer::getSupportedTypes(HashSet<String>& types)
{
types = mimeTypeCache();
}
MediaPlayer::SupportsType MediaPlayerPrivateGStreamer::supportsType(const String& type, const String& codecs)
{
if (type.isNull() || type.isEmpty())
return MediaPlayer::IsNotSupported;
// spec says we should not return "probably" if the codecs string is empty
if (mimeTypeCache().contains(type))
return codecs.isEmpty() ? MediaPlayer::MayBeSupported : MediaPlayer::IsSupported;
return MediaPlayer::IsNotSupported;
}
bool MediaPlayerPrivateGStreamer::hasSingleSecurityOrigin() const
{
return true;
}
bool MediaPlayerPrivateGStreamer::supportsFullscreen() const
{
return true;
}
PlatformMedia MediaPlayerPrivateGStreamer::platformMedia() const
{
PlatformMedia p;
p.type = PlatformMedia::GStreamerGWorldType;
p.media.gstreamerGWorld = m_gstGWorld.get();
return p;
}
void MediaPlayerPrivateGStreamer::setPreload(MediaPlayer::Preload preload)
{
ASSERT(m_playBin);
m_preload = preload;
GstPlayFlags flags;
g_object_get(m_playBin, "flags", &flags, NULL);
if (preload == MediaPlayer::None)
g_object_set(m_playBin, "flags", flags & ~GST_PLAY_FLAG_DOWNLOAD, NULL);
else
g_object_set(m_playBin, "flags", flags | GST_PLAY_FLAG_DOWNLOAD, NULL);
if (m_delayingLoad && m_preload != MediaPlayer::None) {
m_delayingLoad = false;
commitLoad();
}
}
void MediaPlayerPrivateGStreamer::createGSTPlayBin()
{
ASSERT(!m_playBin);
m_playBin = gst_element_factory_make("playbin2", "play");
m_gstGWorld = GStreamerGWorld::createGWorld(m_playBin);
GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_playBin));
gst_bus_add_signal_watch(bus);
g_signal_connect(bus, "message", G_CALLBACK(mediaPlayerPrivateMessageCallback), this);
gst_object_unref(bus);
g_signal_connect(m_playBin, "notify::volume", G_CALLBACK(mediaPlayerPrivateVolumeChangedCallback), this);
g_signal_connect(m_playBin, "notify::source", G_CALLBACK(mediaPlayerPrivateSourceChangedCallback), this);
g_signal_connect(m_playBin, "notify::mute", G_CALLBACK(mediaPlayerPrivateMuteChangedCallback), this);
m_webkitVideoSink = webkit_video_sink_new();
g_signal_connect(m_webkitVideoSink, "repaint-requested", G_CALLBACK(mediaPlayerPrivateRepaintCallback), this);
m_videoSinkBin = gst_bin_new("sink");
GstElement* videoTee = gst_element_factory_make("tee", "videoTee");
GstElement* queue = gst_element_factory_make("queue", 0);
// Take ownership.
g_object_ref_sink(m_videoSinkBin);
// Build a new video sink consisting of a bin containing a tee
// (meant to distribute data to multiple video sinks) and our
// internal video sink. For fullscreen we create an autovideosink
// and initially block the data flow towards it and configure it
gst_bin_add_many(GST_BIN(m_videoSinkBin), videoTee, queue, NULL);
// Link a new src pad from tee to queue1.
GstPad* srcPad = gst_element_get_request_pad(videoTee, "src%d");
GstPad* sinkPad = gst_element_get_static_pad(queue, "sink");
gst_pad_link(srcPad, sinkPad);
gst_object_unref(GST_OBJECT(srcPad));
gst_object_unref(GST_OBJECT(sinkPad));
WTFLogChannel* channel = getChannelFromName("Media");
if (channel->state == WTFLogChannelOn) {
m_fpsSink = gst_element_factory_make("fpsdisplaysink", "sink");
if (g_object_class_find_property(G_OBJECT_GET_CLASS(m_fpsSink), "video-sink")) {
g_object_set(m_fpsSink, "video-sink", m_webkitVideoSink, NULL);
gst_bin_add(GST_BIN(m_videoSinkBin), m_fpsSink);
gst_element_link(queue, m_fpsSink);
} else {
m_fpsSink = 0;
gst_bin_add(GST_BIN(m_videoSinkBin), m_webkitVideoSink);
gst_element_link(queue, m_webkitVideoSink);
LOG_VERBOSE(Media, "Can't display FPS statistics, you need gst-plugins-bad >= 0.10.18");
}
} else {
gst_bin_add(GST_BIN(m_videoSinkBin), m_webkitVideoSink);
gst_element_link(queue, m_webkitVideoSink);
}
// Add a ghostpad to the bin so it can proxy to tee.
GstPad* pad = gst_element_get_static_pad(videoTee, "sink");
gst_element_add_pad(m_videoSinkBin, gst_ghost_pad_new("sink", pad));
gst_object_unref(GST_OBJECT(pad));
// Set the bin as video sink of playbin.
g_object_set(m_playBin, "video-sink", m_videoSinkBin, NULL);
}
}
#endif
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