mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-15 04:46:32 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-good/sys/directsound/gstdirectsoundsink.c

1050 lines
33 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 2005 Sebastien Moutte <sebastien@moutte.net>
* Copyright (C) 2007 Pioneers of the Inevitable <songbird@songbirdnest.com>
* Copyright (C) 2010 Fluendo S.A. <support@fluendo.com>
*
* gstdirectsoundsink.c:
*
* 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 along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*
*
* The development of this code was made possible due to the involvement
* of Pioneers of the Inevitable, the creators of the Songbird Music player
*
*/
/**
* SECTION:element-directsoundsink
* @title: directsoundsink
*
* This element lets you output sound using the DirectSound API.
*
* Note that you should almost always use generic audio conversion elements
* like audioconvert and audioresample in front of an audiosink to make sure
* your pipeline works under all circumstances (those conversion elements will
* act in passthrough-mode if no conversion is necessary).
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! directsoundsink
* ]| will output a sine wave (continuous beep sound) to your sound card (with
* a very low volume as precaution).
* |[
* gst-launch-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! directsoundsink
* ]| will play an Ogg/Vorbis audio file and output it.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/base/gstbasesink.h>
#include "gstdirectsoundsink.h"
#include <gst/audio/gstaudioiec61937.h>
#include <math.h>
#ifdef __CYGWIN__
#include <unistd.h>
#ifndef _swab
#define _swab swab
#endif
#endif
#define DEFAULT_MUTE FALSE
GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
#define GST_CAT_DEFAULT directsoundsink_debug
static void gst_directsound_sink_finalize (GObject * object);
static void gst_directsound_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_directsound_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink,
GstCaps * filter);
static GstBuffer *gst_directsound_sink_payload (GstAudioBaseSink * sink,
GstBuffer * buf);
static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
GstAudioRingBufferSpec * spec);
static gboolean gst_directsound_sink_unprepare (GstAudioSink * asink);
static gboolean gst_directsound_sink_open (GstAudioSink * asink);
static gboolean gst_directsound_sink_close (GstAudioSink * asink);
static gint gst_directsound_sink_write (GstAudioSink * asink,
gpointer data, guint length);
static guint gst_directsound_sink_delay (GstAudioSink * asink);
static void gst_directsound_sink_reset (GstAudioSink * asink);
static GstCaps *gst_directsound_probe_supported_formats (GstDirectSoundSink *
dsoundsink, const GstCaps * template_caps);
static gboolean gst_directsound_sink_query (GstBaseSink * pad,
GstQuery * query);
static void gst_directsound_sink_set_volume (GstDirectSoundSink * sink,
gdouble volume, gboolean store);
static gdouble gst_directsound_sink_get_volume (GstDirectSoundSink * sink);
static void gst_directsound_sink_set_mute (GstDirectSoundSink * sink,
gboolean mute);
static gboolean gst_directsound_sink_get_mute (GstDirectSoundSink * sink);
static const gchar *gst_directsound_sink_get_device (GstDirectSoundSink *
dsoundsink);
static void gst_directsound_sink_set_device (GstDirectSoundSink * dsoundsink,
const gchar * device_id);
static gboolean gst_directsound_sink_is_spdif_format (GstAudioRingBufferSpec *
spec);
static gchar *gst_hres_to_string (HRESULT hRes);
static GstStaticPadTemplate directsoundsink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_DIRECTSOUND_SINK_CAPS));
enum
{
PROP_0,
PROP_VOLUME,
PROP_MUTE,
PROP_DEVICE
};
#define gst_directsound_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstDirectSoundSink, gst_directsound_sink,
GST_TYPE_AUDIO_SINK, G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL)
);
static void
gst_directsound_sink_finalize (GObject * object)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (object);
g_free (dsoundsink->device_id);
dsoundsink->device_id = NULL;
g_mutex_clear (&dsoundsink->dsound_lock);
gst_object_unref (dsoundsink->system_clock);
if (dsoundsink->write_wait_clock_id != NULL) {
gst_clock_id_unref (dsoundsink->write_wait_clock_id);
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
GstAudioBaseSinkClass *gstaudiobasesink_class =
GST_AUDIO_BASE_SINK_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
"DirectSound sink");
gobject_class->finalize = gst_directsound_sink_finalize;
gobject_class->set_property = gst_directsound_sink_set_property;
gobject_class->get_property = gst_directsound_sink_get_property;
gstbasesink_class->get_caps =
GST_DEBUG_FUNCPTR (gst_directsound_sink_getcaps);
gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_directsound_sink_query);
gstaudiobasesink_class->payload =
GST_DEBUG_FUNCPTR (gst_directsound_sink_payload);
gstaudiosink_class->prepare =
GST_DEBUG_FUNCPTR (gst_directsound_sink_prepare);
gstaudiosink_class->unprepare =
GST_DEBUG_FUNCPTR (gst_directsound_sink_unprepare);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_directsound_sink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_directsound_sink_close);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_directsound_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_sink_reset);
g_object_class_install_property (gobject_class,
PROP_VOLUME,
g_param_spec_double ("volume", "Volume",
"Volume of this stream", 0.0, 1.0, 1.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MUTE,
g_param_spec_boolean ("mute", "Mute",
"Mute state of this stream", DEFAULT_MUTE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_DEVICE,
g_param_spec_string ("device", "Device",
"DirectSound playback device as a GUID string",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_static_metadata (element_class,
"Direct Sound Audio Sink", "Sink/Audio",
"Output to a sound card via Direct Sound",
"Sebastien Moutte <sebastien@moutte.net>");
gst_element_class_add_static_pad_template (element_class,
&directsoundsink_sink_factory);
}
static void
gst_directsound_sink_init (GstDirectSoundSink * dsoundsink)
{
dsoundsink->volume = 100;
dsoundsink->mute = FALSE;
dsoundsink->device_id = NULL;
dsoundsink->pDS = NULL;
dsoundsink->cached_caps = NULL;
dsoundsink->pDSBSecondary = NULL;
dsoundsink->current_circular_offset = 0;
dsoundsink->buffer_size = DSBSIZE_MIN;
g_mutex_init (&dsoundsink->dsound_lock);
dsoundsink->system_clock = gst_system_clock_obtain ();
dsoundsink->write_wait_clock_id = NULL;
dsoundsink->first_buffer_after_reset = FALSE;
}
static void
gst_directsound_sink_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object);
switch (prop_id) {
case PROP_VOLUME:
gst_directsound_sink_set_volume (sink, g_value_get_double (value), TRUE);
break;
case PROP_MUTE:
gst_directsound_sink_set_mute (sink, g_value_get_boolean (value));
break;
case PROP_DEVICE:
gst_directsound_sink_set_device (sink, g_value_get_string (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_directsound_sink_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object);
switch (prop_id) {
case PROP_VOLUME:
g_value_set_double (value, gst_directsound_sink_get_volume (sink));
break;
case PROP_MUTE:
g_value_set_boolean (value, gst_directsound_sink_get_mute (sink));
break;
case PROP_DEVICE:
g_value_set_string (value, gst_directsound_sink_get_device (sink));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_directsound_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
{
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (bsink);
GstCaps *caps;
if (dsoundsink->pDS == NULL) {
GST_DEBUG_OBJECT (dsoundsink, "device not open, using template caps");
return NULL; /* base class will get template caps for us */
}
if (dsoundsink->cached_caps) {
caps = gst_caps_ref (dsoundsink->cached_caps);
} else {
element_class = GST_ELEMENT_GET_CLASS (dsoundsink);
pad_template = gst_element_class_get_pad_template (element_class, "sink");
g_return_val_if_fail (pad_template != NULL, NULL);
caps = gst_directsound_probe_supported_formats (dsoundsink,
gst_pad_template_get_caps (pad_template));
if (caps)
dsoundsink->cached_caps = gst_caps_ref (caps);
}
if (caps && filter) {
GstCaps *tmp =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
if (caps) {
gchar *caps_string = gst_caps_to_string (caps);
GST_DEBUG_OBJECT (dsoundsink, "returning caps %s", caps_string);
g_free (caps_string);
}
return caps;
}
static gboolean
gst_directsound_sink_acceptcaps (GstBaseSink * sink, GstQuery * query)
{
GstDirectSoundSink *dsink = GST_DIRECTSOUND_SINK (sink);
GstPad *pad;
GstCaps *caps;
GstCaps *pad_caps;
GstStructure *st;
gboolean ret = FALSE;
GstAudioRingBufferSpec spec = { 0 };
if (G_UNLIKELY (dsink == NULL))
return FALSE;
pad = sink->sinkpad;
gst_query_parse_accept_caps (query, &caps);
GST_DEBUG_OBJECT (pad, "caps %" GST_PTR_FORMAT, caps);
pad_caps = gst_pad_query_caps (pad, NULL);
if (pad_caps) {
gboolean cret = gst_caps_is_subset (caps, pad_caps);
gst_caps_unref (pad_caps);
if (!cret) {
GST_DEBUG_OBJECT (dsink,
"Caps are not a subset of the pad caps, not accepting caps");
goto done;
}
}
/* If we've not got fixed caps, creating a stream might fail, so let's just
* return from here with default acceptcaps behaviour */
if (!gst_caps_is_fixed (caps)) {
GST_DEBUG_OBJECT (dsink, "Caps are not fixed, not accepting caps");
goto done;
}
spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&spec, caps)) {
GST_DEBUG_OBJECT (dsink, "Failed to parse caps, not accepting");
goto done;
}
/* Make sure input is framed (one frame per buffer) and can be payloaded */
switch (spec.type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
{
gboolean framed = FALSE, parsed = FALSE;
st = gst_caps_get_structure (caps, 0);
gst_structure_get_boolean (st, "framed", &framed);
gst_structure_get_boolean (st, "parsed", &parsed);
if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0) {
GST_DEBUG_OBJECT (dsink, "Wrong AC3/DTS caps, not accepting");
goto done;
}
}
default:
break;
}
ret = TRUE;
GST_DEBUG_OBJECT (dsink, "Accepting caps");
done:
gst_query_set_accept_caps_result (query, ret);
return TRUE;
}
static gboolean
gst_directsound_sink_query (GstBaseSink * sink, GstQuery * query)
{
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_ACCEPT_CAPS:
res = gst_directsound_sink_acceptcaps (sink, query);
break;
default:
res = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
}
return res;
}
static LPGUID
string_to_guid (const gchar * str)
{
HRESULT ret;
gunichar2 *wstr;
LPGUID out;
wstr = g_utf8_to_utf16 (str, -1, NULL, NULL, NULL);
if (!wstr)
return NULL;
out = g_new (GUID, 1);
ret = CLSIDFromString ((LPOLESTR) wstr, out);
g_free (wstr);
if (ret != NOERROR) {
g_free (out);
return NULL;
}
return out;
}
static gboolean
gst_directsound_sink_open (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink;
HRESULT hRes;
LPGUID lpGuid = NULL;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
if (dsoundsink->device_id) {
lpGuid = string_to_guid (dsoundsink->device_id);
if (lpGuid == NULL) {
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
("device set but guid not found: %s", dsoundsink->device_id), (NULL));
return FALSE;
}
}
/* create and initialize a DirectSound object */
if (FAILED (hRes = DirectSoundCreate (lpGuid, &dsoundsink->pDS, NULL))) {
gchar *error_text = gst_hres_to_string (hRes);
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
("DirectSoundCreate: %s", error_text), (NULL));
g_free (lpGuid);
g_free (error_text);
return FALSE;
}
g_free (lpGuid);
if (FAILED (hRes = IDirectSound_SetCooperativeLevel (dsoundsink->pDS,
GetDesktopWindow (), DSSCL_PRIORITY))) {
gchar *error_text = gst_hres_to_string (hRes);
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
("IDirectSound_SetCooperativeLevel: %s", error_text), (NULL));
g_free (error_text);
return FALSE;
}
return TRUE;
}
static gboolean
gst_directsound_sink_is_spdif_format (GstAudioRingBufferSpec * spec)
{
return spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3 ||
spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS;
}
static gboolean
gst_directsound_sink_prepare (GstAudioSink * asink,
GstAudioRingBufferSpec * spec)
{
GstDirectSoundSink *dsoundsink;
HRESULT hRes;
DSBUFFERDESC descSecondary;
WAVEFORMATEX wfx;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/*save number of bytes per sample and buffer format */
dsoundsink->bytes_per_sample = spec->info.bpf;
dsoundsink->type = spec->type;
/* fill the WAVEFORMATEX structure with spec params */
memset (&wfx, 0, sizeof (wfx));
if (!gst_directsound_sink_is_spdif_format (spec)) {
wfx.cbSize = sizeof (wfx);
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = spec->info.channels;
wfx.nSamplesPerSec = spec->info.rate;
wfx.wBitsPerSample = (spec->info.bpf * 8) / wfx.nChannels;
wfx.nBlockAlign = spec->info.bpf;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
/* Create directsound buffer with size based on our configured
* buffer_size (which is 200 ms by default) */
dsoundsink->buffer_size =
gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->buffer_time,
GST_MSECOND);
/* Make sure we make those numbers multiple of our sample size in bytes */
dsoundsink->buffer_size -= dsoundsink->buffer_size % spec->info.bpf;
spec->segsize =
gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
GST_MSECOND);
spec->segsize -= spec->segsize % spec->info.bpf;
spec->segtotal = dsoundsink->buffer_size / spec->segsize;
} else {
#ifdef WAVE_FORMAT_DOLBY_AC3_SPDIF
wfx.cbSize = 0;
wfx.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
wfx.nChannels = 2;
wfx.nSamplesPerSec = 48000;
wfx.wBitsPerSample = 16;
wfx.nBlockAlign = wfx.wBitsPerSample / 8 * wfx.nChannels;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
spec->segsize = 6144;
spec->segtotal = 10;
#else
g_assert_not_reached ();
#endif
}
// Make the final buffer size be an integer number of segments
dsoundsink->buffer_size = spec->segsize * spec->segtotal;
GST_INFO_OBJECT (dsoundsink, "channels: %d, rate: %d, bytes_per_sample: %d"
" WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
" WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n"
"Size of dsound circular buffer=>%d\n",
GST_AUDIO_INFO_CHANNELS (&spec->info), GST_AUDIO_INFO_RATE (&spec->info),
GST_AUDIO_INFO_BPF (&spec->info), wfx.nSamplesPerSec, wfx.wBitsPerSample,
wfx.nBlockAlign, wfx.nAvgBytesPerSec, dsoundsink->buffer_size);
/* create a secondary directsound buffer */
memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
descSecondary.dwSize = sizeof (DSBUFFERDESC);
descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
if (!gst_directsound_sink_is_spdif_format (spec))
descSecondary.dwFlags |= DSBCAPS_CTRLVOLUME;
descSecondary.dwBufferBytes = dsoundsink->buffer_size;
descSecondary.lpwfxFormat = (WAVEFORMATEX *) & wfx;
hRes = IDirectSound_CreateSoundBuffer (dsoundsink->pDS, &descSecondary,
&dsoundsink->pDSBSecondary, NULL);
if (FAILED (hRes)) {
gchar *error_text = gst_hres_to_string (hRes);
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
("IDirectSound_CreateSoundBuffer: %s", error_text), (NULL));
g_free (error_text);
return FALSE;
}
gst_directsound_sink_set_volume (dsoundsink,
gst_directsound_sink_get_volume (dsoundsink), FALSE);
gst_directsound_sink_set_mute (dsoundsink, dsoundsink->mute);
return TRUE;
}
static gboolean
gst_directsound_sink_unprepare (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/* release secondary DirectSound buffer */
if (dsoundsink->pDSBSecondary) {
IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
dsoundsink->pDSBSecondary = NULL;
}
return TRUE;
}
static gboolean
gst_directsound_sink_close (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink = NULL;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/* release DirectSound object */
g_return_val_if_fail (dsoundsink->pDS != NULL, FALSE);
IDirectSound_Release (dsoundsink->pDS);
dsoundsink->pDS = NULL;
gst_caps_replace (&dsoundsink->cached_caps, NULL);
return TRUE;
}
static gint
gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstDirectSoundSink *dsoundsink;
DWORD dwStatus = 0;
HRESULT hRes, hRes2;
LPVOID pLockedBuffer1 = NULL, pLockedBuffer2 = NULL;
DWORD dwSizeBuffer1, dwSizeBuffer2;
DWORD dwCurrentPlayCursor;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
GST_DSOUND_LOCK (dsoundsink);
/* get current buffer status */
hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
/* get current play cursor position */
hRes2 = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
&dwCurrentPlayCursor, NULL);
if (SUCCEEDED (hRes) && SUCCEEDED (hRes2) && (dwStatus & DSBSTATUS_PLAYING)) {
DWORD dwFreeBufferSize = 0;
GstClockTime sleep_time_ms = 0, sleep_until;
GstClockID clock_id;
calculate_freesize:
/* Calculate the free space in the circular buffer */
if (dwCurrentPlayCursor < dsoundsink->current_circular_offset)
dwFreeBufferSize =
dsoundsink->buffer_size - (dsoundsink->current_circular_offset -
dwCurrentPlayCursor);
else
dwFreeBufferSize =
dwCurrentPlayCursor - dsoundsink->current_circular_offset;
/* Not enough free space, wait for some samples to be played out. We could
* write out partial data, but that will result in a tight loop in the
* audioringbuffer write thread, and lead to high CPU usage. */
if (length > dwFreeBufferSize) {
gint rate = GST_AUDIO_BASE_SINK (asink)->ringbuffer->spec.info.rate;
/* Wait for a time proportional to the space needed. In reality, the
* directsound sink's position does not update frequently enough, so we
* will end up waiting for much longer. Note that Sleep() has millisecond
* resolution at best. */
sleep_time_ms = gst_util_uint64_scale_int ((length - dwFreeBufferSize),
1000, dsoundsink->bytes_per_sample * rate);
/* Make sure we don't run in a tight loop unnecessarily */
sleep_time_ms = MAX (sleep_time_ms, 10);
sleep_until = gst_clock_get_time (dsoundsink->system_clock) +
sleep_time_ms * GST_MSECOND;
GST_DEBUG_OBJECT (dsoundsink,
"length: %u, FreeBufSiz: %ld, sleep_time_ms: %" G_GUINT64_FORMAT
", bps: %i, rate: %i", length, dwFreeBufferSize, sleep_time_ms,
dsoundsink->bytes_per_sample, rate);
if (G_UNLIKELY (dsoundsink->write_wait_clock_id == NULL ||
gst_clock_single_shot_id_reinit (dsoundsink->system_clock,
dsoundsink->write_wait_clock_id, sleep_until) == FALSE)) {
if (dsoundsink->write_wait_clock_id != NULL) {
gst_clock_id_unref (dsoundsink->write_wait_clock_id);
}
dsoundsink->write_wait_clock_id =
gst_clock_new_single_shot_id (dsoundsink->system_clock,
sleep_until);
}
clock_id = dsoundsink->write_wait_clock_id;
dsoundsink->reset_while_sleeping = FALSE;
GST_DSOUND_UNLOCK (dsoundsink);
/* don't bother with the return value as we'll detect reset separately,
as reset could happen between when this returns and we obtain the lock
again -- so we can't use UNSCHEDULED here */
gst_clock_id_wait (clock_id, NULL);
GST_DSOUND_LOCK (dsoundsink);
/* if a reset occurs, exit now */
if (dsoundsink->reset_while_sleeping == TRUE) {
GST_DSOUND_UNLOCK (dsoundsink);
return -1;
}
/* May we send out? */
hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
&dwCurrentPlayCursor, NULL);
hRes2 =
IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
if (SUCCEEDED (hRes) && SUCCEEDED (hRes2)
&& (dwStatus & DSBSTATUS_PLAYING))
goto calculate_freesize;
else {
gchar *err1, *err2;
dsoundsink->first_buffer_after_reset = FALSE;
GST_DSOUND_UNLOCK (dsoundsink);
err1 = gst_hres_to_string (hRes);
err2 = gst_hres_to_string (hRes2);
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_WRITE,
("IDirectSoundBuffer_GetStatus %s, "
"IDirectSoundBuffer_GetCurrentPosition: %s, dwStatus: %lu",
err2, err1, dwStatus), (NULL));
g_free (err1);
g_free (err2);
return -1;
}
}
}
if (dwStatus & DSBSTATUS_BUFFERLOST) {
hRes = IDirectSoundBuffer_Restore (dsoundsink->pDSBSecondary); /*need a loop waiting the buffer is restored?? */
dsoundsink->current_circular_offset = 0;
}
/* Lock a buffer of length @length for writing */
hRes = IDirectSoundBuffer_Lock (dsoundsink->pDSBSecondary,
dsoundsink->current_circular_offset, length, &pLockedBuffer1,
&dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
if (SUCCEEDED (hRes)) {
// Write to pointers without reordering.
memcpy (pLockedBuffer1, data, dwSizeBuffer1);
if (pLockedBuffer2 != NULL)
memcpy (pLockedBuffer2, (LPBYTE) data + dwSizeBuffer1, dwSizeBuffer2);
hRes = IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer1,
dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
// Update where the buffer will lock (for next time)
dsoundsink->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
dsoundsink->current_circular_offset %= dsoundsink->buffer_size; /* Circular buffer */
}
/* if the buffer was not in playing state yet, call play on the buffer
except if this buffer is the fist after a reset (base class call reset and write a buffer when setting the sink to pause) */
if (!(dwStatus & DSBSTATUS_PLAYING) &&
dsoundsink->first_buffer_after_reset == FALSE) {
hRes = IDirectSoundBuffer_Play (dsoundsink->pDSBSecondary, 0, 0,
DSBPLAY_LOOPING);
}
dsoundsink->first_buffer_after_reset = FALSE;
GST_DSOUND_UNLOCK (dsoundsink);
return length;
}
static guint
gst_directsound_sink_delay (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink;
HRESULT hRes;
DWORD dwCurrentPlayCursor;
DWORD dwBytesInQueue = 0;
gint nNbSamplesInQueue = 0;
DWORD dwStatus;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/* get current buffer status */
hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
if (SUCCEEDED (hRes) && (dwStatus & DSBSTATUS_PLAYING)) {
/*evaluate the number of samples in queue in the circular buffer */
hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
&dwCurrentPlayCursor, NULL);
if (hRes == S_OK) {
if (dwCurrentPlayCursor < dsoundsink->current_circular_offset)
dwBytesInQueue =
dsoundsink->current_circular_offset - dwCurrentPlayCursor;
else
dwBytesInQueue =
dsoundsink->current_circular_offset + (dsoundsink->buffer_size -
dwCurrentPlayCursor);
nNbSamplesInQueue = dwBytesInQueue / dsoundsink->bytes_per_sample;
}
}
return nNbSamplesInQueue;
}
static void
gst_directsound_sink_reset (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink;
LPVOID pLockedBuffer = NULL;
DWORD dwSizeBuffer = 0;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
GST_DSOUND_LOCK (dsoundsink);
if (dsoundsink->pDSBSecondary) {
/*stop playing */
HRESULT hRes = IDirectSoundBuffer_Stop (dsoundsink->pDSBSecondary);
/*reset position */
hRes = IDirectSoundBuffer_SetCurrentPosition (dsoundsink->pDSBSecondary, 0);
dsoundsink->current_circular_offset = 0;
/*reset the buffer */
hRes = IDirectSoundBuffer_Lock (dsoundsink->pDSBSecondary,
0, dsoundsink->buffer_size,
&pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L);
if (SUCCEEDED (hRes)) {
memset (pLockedBuffer, 0, dwSizeBuffer);
hRes =
IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer,
dwSizeBuffer, NULL, 0);
}
}
dsoundsink->reset_while_sleeping = TRUE;
dsoundsink->first_buffer_after_reset = TRUE;
if (dsoundsink->write_wait_clock_id != NULL) {
gst_clock_id_unschedule (dsoundsink->write_wait_clock_id);
}
GST_DSOUND_UNLOCK (dsoundsink);
}
/*
* gst_directsound_probe_supported_formats:
*
* Takes the template caps and returns the subset which is actually
* supported by this device.
*
*/
static GstCaps *
gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
const GstCaps * template_caps)
{
HRESULT hRes;
DSBUFFERDESC descSecondary;
WAVEFORMATEX wfx;
GstCaps *caps;
GstCaps *tmp, *tmp2;
LPDIRECTSOUNDBUFFER tmpBuffer;
caps = gst_caps_copy (template_caps);
/*
* Check availability of digital output by trying to create an SPDIF buffer
*/
#ifdef WAVE_FORMAT_DOLBY_AC3_SPDIF
/* fill the WAVEFORMATEX structure with some standard AC3 over SPDIF params */
memset (&wfx, 0, sizeof (wfx));
wfx.cbSize = 0;
wfx.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
wfx.nChannels = 2;
wfx.nSamplesPerSec = 48000;
wfx.wBitsPerSample = 16;
wfx.nBlockAlign = 4;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
// create a secondary directsound buffer
memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
descSecondary.dwSize = sizeof (DSBUFFERDESC);
descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
descSecondary.dwBufferBytes = 6144;
descSecondary.lpwfxFormat = &wfx;
hRes = IDirectSound_CreateSoundBuffer (dsoundsink->pDS, &descSecondary,
&tmpBuffer, NULL);
if (FAILED (hRes)) {
gchar *error_text = gst_hres_to_string (hRes);
GST_INFO_OBJECT (dsoundsink, "AC3 passthrough not supported "
"(IDirectSound_CreateSoundBuffer returned: %s)\n", error_text);
g_free (error_text);
tmp = gst_caps_new_empty_simple ("audio/x-ac3");
tmp2 = gst_caps_subtract (caps, tmp);
gst_caps_unref (tmp);
gst_caps_unref (caps);
caps = tmp2;
tmp = gst_caps_new_empty_simple ("audio/x-dts");
tmp2 = gst_caps_subtract (caps, tmp);
gst_caps_unref (tmp);
gst_caps_unref (caps);
caps = tmp2;
} else {
GST_INFO_OBJECT (dsoundsink, "AC3 passthrough supported");
hRes = IDirectSoundBuffer_Release (tmpBuffer);
if (FAILED (hRes)) {
gchar *error_text = gst_hres_to_string (hRes);
GST_DEBUG_OBJECT (dsoundsink,
"(IDirectSoundBuffer_Release returned: %s)\n", error_text);
g_free (error_text);
}
}
#else
tmp = gst_caps_new_empty_simple ("audio/x-ac3");
tmp2 = gst_caps_subtract (caps, tmp);
gst_caps_unref (tmp);
gst_caps_unref (caps);
caps = tmp2;
tmp = gst_caps_new_empty_simple ("audio/x-dts");
tmp2 = gst_caps_subtract (caps, tmp);
gst_caps_unref (tmp);
gst_caps_unref (caps);
caps = tmp2;
#endif
return caps;
}
static GstBuffer *
gst_directsound_sink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
{
if (gst_directsound_sink_is_spdif_format (&sink->ringbuffer->spec)) {
gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
GstBuffer *out;
GstMapInfo infobuf, infoout;
gboolean success;
if (framesize <= 0)
return NULL;
out = gst_buffer_new_and_alloc (framesize);
if (!gst_buffer_map (buf, &infobuf, GST_MAP_READWRITE)) {
gst_buffer_unref (out);
return NULL;
}
if (!gst_buffer_map (out, &infoout, GST_MAP_READWRITE)) {
gst_buffer_unmap (buf, &infobuf);
gst_buffer_unref (out);
return NULL;
}
success = gst_audio_iec61937_payload (infobuf.data, infobuf.size,
infoout.data, infoout.size, &sink->ringbuffer->spec, G_BYTE_ORDER);
if (!success) {
gst_buffer_unmap (out, &infoout);
gst_buffer_unmap (buf, &infobuf);
gst_buffer_unref (out);
return NULL;
}
gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_ALL, 0, -1);
/* Fix endianness */
_swab ((gchar *) infoout.data, (gchar *) infoout.data, infobuf.size);
gst_buffer_unmap (out, &infoout);
gst_buffer_unmap (buf, &infobuf);
return out;
} else
return gst_buffer_ref (buf);
}
static void
gst_directsound_sink_set_volume (GstDirectSoundSink * dsoundsink,
gdouble dvolume, gboolean store)
{
glong volume;
volume = dvolume * 100;
if (store)
dsoundsink->volume = volume;
if (dsoundsink->pDSBSecondary) {
/* DirectSound controls volume using units of 100th of a decibel,
* ranging from -10000 to 0. We use a linear scale of 0 - 100
* here, so remap.
*/
long dsVolume;
if (volume == 0 || dsoundsink->mute)
dsVolume = -10000;
else
dsVolume = 100 * (long) (20 * log10 ((double) volume / 100.));
dsVolume = CLAMP (dsVolume, -10000, 0);
GST_DEBUG_OBJECT (dsoundsink,
"Setting volume on secondary buffer to %d from %d", (int) dsVolume,
(int) volume);
IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary, dsVolume);
}
}
gdouble
gst_directsound_sink_get_volume (GstDirectSoundSink * dsoundsink)
{
return (gdouble) dsoundsink->volume / 100;
}
static void
gst_directsound_sink_set_mute (GstDirectSoundSink * dsoundsink, gboolean mute)
{
if (mute) {
gst_directsound_sink_set_volume (dsoundsink, 0, FALSE);
dsoundsink->mute = TRUE;
} else {
gst_directsound_sink_set_volume (dsoundsink,
gst_directsound_sink_get_volume (dsoundsink), FALSE);
dsoundsink->mute = FALSE;
}
}
static gboolean
gst_directsound_sink_get_mute (GstDirectSoundSink * dsoundsink)
{
return dsoundsink->mute;
}
static const gchar *
gst_directsound_sink_get_device (GstDirectSoundSink * dsoundsink)
{
return dsoundsink->device_id;
}
static void
gst_directsound_sink_set_device (GstDirectSoundSink * dsoundsink,
const gchar * device_id)
{
g_free (dsoundsink->device_id);
dsoundsink->device_id = g_strdup (device_id);
}
/* Converts a HRESULT error to a text string
* LPTSTR is either a */
static gchar *
gst_hres_to_string (HRESULT hRes)
{
DWORD flags;
gchar *ret_text;
LPTSTR error_text = NULL;
flags = FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_ALLOCATE_BUFFER
| FORMAT_MESSAGE_IGNORE_INSERTS;
FormatMessage (flags, NULL, hRes, MAKELANGID (LANG_NEUTRAL, SUBLANG_DEFAULT),
(LPTSTR) & error_text, 0, NULL);
#ifdef UNICODE
/* If UNICODE is defined, LPTSTR is LPWSTR which is UTF-16 */
ret_text = g_utf16_to_utf8 (error_text, 0, NULL, NULL, NULL);
#else
ret_text = g_strdup (error_text);
#endif
LocalFree (error_text);
return ret_text;
}
Reference in a new issue View git blame Copy permalink