diff options
Diffstat (limited to 'src/pulsesrc.c')
| -rw-r--r-- | src/pulsesrc.c | 616 |
1 files changed, 616 insertions, 0 deletions
diff --git a/src/pulsesrc.c b/src/pulsesrc.c new file mode 100644 index 0000000..c830ca8 --- /dev/null +++ b/src/pulsesrc.c @@ -0,0 +1,616 @@ +/* $Id$ */ + +/*** + This file is part of gst-pulse. + + gst-pulse is free software; you can redistribute it and/or modify + it under the terms of the GNU Lesser General Public License as + published by the Free Software Foundation; either version 2.1 of the + License, or (at your option) any later version. + + gst-pulse 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 + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with gst-pulse; if not, write to the Free Software + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 + USA. +***/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <string.h> +#include <stdio.h> + +#include <gst/base/gstbasesrc.h> +#include <gst/gsttaglist.h> + +#include "pulsesrc.h" +#include "pulseutil.h" +#include "pulsemixerctrl.h" + +GST_DEBUG_CATEGORY_EXTERN(pulse_debug); +#define GST_CAT_DEFAULT pulse_debug + +enum { + PROP_SERVER = 1, + PROP_DEVICE +}; + +static GstAudioSrcClass *parent_class = NULL; + +GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS(GstPolypSrc, gst_pulsesrc) + +static void gst_pulsesrc_destroy_stream(GstPolypSrc *pulsesrc); +static void gst_pulsesrc_destroy_context(GstPolypSrc *pulsesrc); + +static void gst_pulsesrc_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec); +static void gst_pulsesrc_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec); +static void gst_pulsesrc_finalize(GObject *object); +static void gst_pulsesrc_dispose(GObject *object); + +static gboolean gst_pulsesrc_open(GstAudioSrc *asrc); +static gboolean gst_pulsesrc_close(GstAudioSrc *asrc); + +static gboolean gst_pulsesrc_prepare(GstAudioSrc *asrc, GstRingBufferSpec *spec); +static gboolean gst_pulsesrc_unprepare(GstAudioSrc *asrc); + +static guint gst_pulsesrc_read(GstAudioSrc *asrc, gpointer data, guint length); +static guint gst_pulsesrc_delay(GstAudioSrc *asrc); + +static GstStateChangeReturn gst_pulsesrc_change_state(GstElement *element, GstStateChange transition); + +#if (G_BYTE_ORDER == G_LITTLE_ENDIAN) +# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN" +#else +# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN" +#endif + +static gboolean gst_pulsesrc_interface_supported(GstImplementsInterface* iface, GType interface_type) { + GstPolypSrc *this = GST_PULSESRC(iface); + + if (interface_type == GST_TYPE_MIXER && this->mixer) + return TRUE; + + return FALSE; +} + +static void gst_pulsesrc_implements_interface_init(GstImplementsInterfaceClass* klass) { + klass->supported = gst_pulsesrc_interface_supported; +} + +static void gst_pulsesrc_init_interfaces(GType type) { + static const GInterfaceInfo implements_iface_info = { + (GInterfaceInitFunc) gst_pulsesrc_implements_interface_init, + NULL, + NULL, + }; + static const GInterfaceInfo mixer_iface_info = { + (GInterfaceInitFunc) gst_pulsesrc_mixer_interface_init, + NULL, + NULL, + }; + + g_type_add_interface_static(type, GST_TYPE_IMPLEMENTS_INTERFACE, &implements_iface_info); + g_type_add_interface_static(type, GST_TYPE_MIXER, &mixer_iface_info); +} + +static void gst_pulsesrc_base_init(gpointer g_class) { + + static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE( + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS( + "audio/x-raw-int, " + "endianness = (int) { " ENDIANNESS " }, " + "signed = (boolean) TRUE, " + "width = (int) 16, " + "depth = (int) 16, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, 16 ];" + + "audio/x-raw-int, " + "signed = (boolean) FALSE, " + "width = (int) 8, " + "depth = (int) 8, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, 16 ];" + + "audio/x-raw-float, " + "endianness = (int) { " ENDIANNESS " }, " + "width = (int) 32, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, 16 ];" + + "audio/x-alaw, " + "rate = (int) [ 1, MAX], " + "channels = (int) [ 1, 16 ];" + + "audio/x-mulaw, " + "rate = (int) [ 1, MAX], " + "channels = (int) [ 1, 16 ]" + ) + ); + + static const GstElementDetails details = + GST_ELEMENT_DETAILS( + "PulseAudio Audio Source", + "Source/Audio", + "Captures audio from a PulseAudio server", + "Lennart Poettering"); + + GstElementClass *element_class = GST_ELEMENT_CLASS(g_class); + + gst_element_class_set_details(element_class, &details); + gst_element_class_add_pad_template(element_class, gst_static_pad_template_get(&pad_template)); +} + +static void gst_pulsesrc_class_init( + gpointer g_class, + gpointer class_data) { + + GObjectClass *gobject_class = G_OBJECT_CLASS(g_class); + GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS(g_class); + GstElementClass *gstelement_class = GST_ELEMENT_CLASS(g_class); + parent_class = g_type_class_peek_parent(g_class); + + gstelement_class->change_state = GST_DEBUG_FUNCPTR(gst_pulsesrc_change_state); + + gobject_class->dispose = GST_DEBUG_FUNCPTR(gst_pulsesrc_dispose); + gobject_class->finalize = GST_DEBUG_FUNCPTR(gst_pulsesrc_finalize); + gobject_class->set_property = GST_DEBUG_FUNCPTR(gst_pulsesrc_set_property); + gobject_class->get_property = GST_DEBUG_FUNCPTR(gst_pulsesrc_get_property); + + gstaudiosrc_class->open = GST_DEBUG_FUNCPTR(gst_pulsesrc_open); + gstaudiosrc_class->close = GST_DEBUG_FUNCPTR(gst_pulsesrc_close); + gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR(gst_pulsesrc_prepare); + gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR(gst_pulsesrc_unprepare); + gstaudiosrc_class->read = GST_DEBUG_FUNCPTR(gst_pulsesrc_read); + gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR(gst_pulsesrc_delay); + + /* Overwrite GObject fields */ + g_object_class_install_property( + gobject_class, + PROP_SERVER, + g_param_spec_string("server", "Server", "The PulseAudio server to connect to", NULL, G_PARAM_READWRITE)); + g_object_class_install_property( + gobject_class, + PROP_DEVICE, + g_param_spec_string("device", "Source", "The PulseAudio source device to connect to", NULL, G_PARAM_READWRITE)); +} + +static void gst_pulsesrc_init( + GTypeInstance * instance, + gpointer g_class) { + + GstPolypSrc *pulsesrc = GST_PULSESRC(instance); + int e; + + pulsesrc->server = pulsesrc->device = NULL; + + pulsesrc->context = NULL; + pulsesrc->stream = NULL; + + pulsesrc->read_buffer = NULL; + pulsesrc->read_buffer_length = 0; + + pulsesrc->mainloop = pa_threaded_mainloop_new(); + g_assert(pulsesrc->mainloop); + + e = pa_threaded_mainloop_start(pulsesrc->mainloop); + g_assert(e == 0); + + pulsesrc->mixer = NULL; +} + +static void gst_pulsesrc_destroy_stream(GstPolypSrc* pulsesrc) { + if (pulsesrc->stream) { + pa_stream_disconnect(pulsesrc->stream); + pa_stream_unref(pulsesrc->stream); + pulsesrc->stream = NULL; + } +} + +static void gst_pulsesrc_destroy_context(GstPolypSrc* pulsesrc) { + + gst_pulsesrc_destroy_stream(pulsesrc); + + if (pulsesrc->context) { + pa_context_disconnect(pulsesrc->context); + pa_context_unref(pulsesrc->context); + pulsesrc->context = NULL; + } +} + +static void gst_pulsesrc_finalize(GObject * object) { + GstPolypSrc *pulsesrc = GST_PULSESRC(object); + + pa_threaded_mainloop_stop(pulsesrc->mainloop); + + gst_pulsesrc_destroy_context(pulsesrc); + + g_free(pulsesrc->server); + g_free(pulsesrc->device); + + pa_threaded_mainloop_free(pulsesrc->mainloop); + + if (pulsesrc->mixer) + gst_pulsemixer_ctrl_free(pulsesrc->mixer); + + G_OBJECT_CLASS(parent_class)->finalize(object); +} + +static void gst_pulsesrc_dispose(GObject * object) { + G_OBJECT_CLASS(parent_class)->dispose(object); +} + +static void gst_pulsesrc_set_property( + GObject * object, + guint prop_id, + const GValue * value, + GParamSpec * pspec) { + + GstPolypSrc *pulsesrc = GST_PULSESRC(object); + + switch (prop_id) { + case PROP_SERVER: + g_free(pulsesrc->server); + pulsesrc->server = g_value_dup_string(value); + break; + + case PROP_DEVICE: + g_free(pulsesrc->device); + pulsesrc->device = g_value_dup_string(value); + break; + + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec); + break; + } +} + +static void gst_pulsesrc_get_property( + GObject * object, + guint prop_id, + GValue * value, + GParamSpec * pspec) { + + GstPolypSrc *pulsesrc = GST_PULSESRC(object); + + switch(prop_id) { + case PROP_SERVER: + g_value_set_string(value, pulsesrc->server); + break; + + case PROP_DEVICE: + g_value_set_string(value, pulsesrc->device); + break; + + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec); + break; + } +} + +static void gst_pulsesrc_context_state_cb(pa_context *c, void *userdata) { + GstPolypSrc *pulsesrc = GST_PULSESRC(userdata); + + switch (pa_context_get_state(c)) { + case PA_CONTEXT_READY: + case PA_CONTEXT_TERMINATED: + case PA_CONTEXT_FAILED: + pa_threaded_mainloop_signal(pulsesrc->mainloop, 0); + break; + + case PA_CONTEXT_UNCONNECTED: + case PA_CONTEXT_CONNECTING: + case PA_CONTEXT_AUTHORIZING: + case PA_CONTEXT_SETTING_NAME: + break; + } +} + +static void gst_pulsesrc_stream_state_cb(pa_stream *s, void * userdata) { + GstPolypSrc *pulsesrc = GST_PULSESRC(userdata); + + switch (pa_stream_get_state(s)) { + + case PA_STREAM_READY: + case PA_STREAM_FAILED: + case PA_STREAM_TERMINATED: + pa_threaded_mainloop_signal(pulsesrc->mainloop, 0); + break; + + case PA_STREAM_UNCONNECTED: + case PA_STREAM_CREATING: + break; + } +} + +static void gst_pulsesrc_stream_request_cb(pa_stream *s, size_t length, void *userdata) { + GstPolypSrc *pulsesrc = GST_PULSESRC(userdata); + + pa_threaded_mainloop_signal(pulsesrc->mainloop, 0); +} + +static gboolean gst_pulsesrc_open(GstAudioSrc *asrc) { + GstPolypSrc *pulsesrc = GST_PULSESRC(asrc); + gchar *name = gst_pulse_client_name(); + + pa_threaded_mainloop_lock(pulsesrc->mainloop); + + if (!(pulsesrc->context = pa_context_new(pa_threaded_mainloop_get_api(pulsesrc->mainloop), name))) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to create context"), (NULL)); + goto unlock_and_fail; + } + + pa_context_set_state_callback(pulsesrc->context, gst_pulsesrc_context_state_cb, pulsesrc); + + if (pa_context_connect(pulsesrc->context, pulsesrc->server, 0, NULL) < 0) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL)); + goto unlock_and_fail; + } + + /* Wait until the context is ready */ + pa_threaded_mainloop_wait(pulsesrc->mainloop); + + if (pa_context_get_state(pulsesrc->context) != PA_CONTEXT_READY) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL)); + goto unlock_and_fail; + } + + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + + g_free(name); + return TRUE; + +unlock_and_fail: + + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + + g_free(name); + return FALSE; +} + +static gboolean gst_pulsesrc_close(GstAudioSrc *asrc) { + GstPolypSrc *pulsesrc = GST_PULSESRC(asrc); + + pa_threaded_mainloop_lock(pulsesrc->mainloop); + gst_pulsesrc_destroy_context(pulsesrc); + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + + return TRUE; +} +static gboolean gst_pulsesrc_prepare(GstAudioSrc *asrc, GstRingBufferSpec *spec) { + pa_buffer_attr buf_attr; + + GstPolypSrc *pulsesrc = GST_PULSESRC(asrc); + + if (!gst_pulse_fill_sample_spec(spec, &pulsesrc->sample_spec)) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, SETTINGS, ("Invalid sample specification."), (NULL)); + goto unlock_and_fail; + } + + pa_threaded_mainloop_lock(pulsesrc->mainloop); + + if (!pulsesrc->context || pa_context_get_state(pulsesrc->context) != PA_CONTEXT_READY) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Bad context state: %s", pulsesrc->context ? pa_strerror(pa_context_errno(pulsesrc->context)) : NULL), (NULL)); + goto unlock_and_fail; + } + + if (!(pulsesrc->stream = pa_stream_new(pulsesrc->context, "Record Stream", &pulsesrc->sample_spec, NULL))) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to create stream: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL)); + goto unlock_and_fail; + } + + pa_stream_set_state_callback(pulsesrc->stream, gst_pulsesrc_stream_state_cb, pulsesrc); + pa_stream_set_read_callback(pulsesrc->stream, gst_pulsesrc_stream_request_cb, pulsesrc); + + memset(&buf_attr, 0, sizeof(buf_attr)); + buf_attr.maxlength = spec->segtotal*spec->segsize*2; + buf_attr.fragsize = spec->segsize; + + if (pa_stream_connect_record(pulsesrc->stream, pulsesrc->device, &buf_attr, PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE|PA_STREAM_NOT_MONOTONOUS) < 0) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to connect stream: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL)); + goto unlock_and_fail; + } + + /* Wait until the stream is ready */ + pa_threaded_mainloop_wait(pulsesrc->mainloop); + + if (pa_stream_get_state(pulsesrc->stream) != PA_STREAM_READY) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to connect stream: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL)); + goto unlock_and_fail; + } + + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + + spec->bytes_per_sample = pa_frame_size(&pulsesrc->sample_spec); + memset(spec->silence_sample, 0, spec->bytes_per_sample); + + return TRUE; + +unlock_and_fail: + + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + return FALSE; +} + +static gboolean gst_pulsesrc_unprepare(GstAudioSrc * asrc) { + GstPolypSrc *pulsesrc = GST_PULSESRC(asrc); + + pa_threaded_mainloop_lock(pulsesrc->mainloop); + gst_pulsesrc_destroy_stream(pulsesrc); + + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + + pulsesrc->read_buffer = NULL; + pulsesrc->read_buffer_length = 0; + + return TRUE; +} + +#define CHECK_DEAD_GOTO(pulsesrc, label) \ +if (!(pulsesrc)->context || pa_context_get_state((pulsesrc)->context) != PA_CONTEXT_READY || \ + !(pulsesrc)->stream || pa_stream_get_state((pulsesrc)->stream) != PA_STREAM_READY) { \ + GST_ELEMENT_ERROR((pulsesrc), RESOURCE, FAILED, ("Disconnected: %s", (pulsesrc)->context ? pa_strerror(pa_context_errno((pulsesrc)->context)) : NULL), (NULL)); \ + goto label; \ +} + +static guint gst_pulsesrc_read(GstAudioSrc *asrc, gpointer data, guint length) { + GstPolypSrc *pulsesrc = GST_PULSESRC(asrc); + size_t sum = 0; + + pa_threaded_mainloop_lock(pulsesrc->mainloop); + + CHECK_DEAD_GOTO(pulsesrc, unlock_and_fail); + + while (length > 0) { + size_t l; + + if (!pulsesrc->read_buffer) { + + for (;;) { + if (pa_stream_peek(pulsesrc->stream, &pulsesrc->read_buffer, &pulsesrc->read_buffer_length) < 0) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("pa_stream_peek() failed: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL)); + goto unlock_and_fail; + } + + if (pulsesrc->read_buffer) + break; + + pa_threaded_mainloop_wait(pulsesrc->mainloop); + + CHECK_DEAD_GOTO(pulsesrc, unlock_and_fail); + } + } + + g_assert(pulsesrc->read_buffer && pulsesrc->read_buffer_length); + + l = pulsesrc->read_buffer_length > length ? length : pulsesrc->read_buffer_length; + + memcpy(data, pulsesrc->read_buffer, l); + + pulsesrc->read_buffer = (const guint8*) pulsesrc->read_buffer + l; + pulsesrc->read_buffer_length -= l; + + data = (guint8*) data + l; + length -= l; + + sum += l; + + if (pulsesrc->read_buffer_length <= 0) { + + if (pa_stream_drop(pulsesrc->stream) < 0) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("pa_stream_drop() failed: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL)); + goto unlock_and_fail; + } + + pulsesrc->read_buffer = NULL; + pulsesrc->read_buffer_length = 0; + } + } + + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + + return sum; + +unlock_and_fail: + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + return 0; +} + +static guint gst_pulsesrc_delay(GstAudioSrc *asrc) { + GstPolypSrc *pulsesrc = GST_PULSESRC(asrc); + pa_usec_t t; + int negative; + + pa_threaded_mainloop_lock(pulsesrc->mainloop); + + CHECK_DEAD_GOTO(pulsesrc, unlock_and_fail); + + if (pa_stream_get_latency(pulsesrc->stream, &t, &negative) < 0) { + + if (pa_context_errno(pulsesrc->context) != PA_ERR_NODATA) { + GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("pa_stream_get_latency() failed: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL)); + goto unlock_and_fail; + } + + GST_WARNING("Not data while querying latency"); + t = 0; + } else if (negative) + t = 0; + + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + + return (guint) ((t * pulsesrc->sample_spec.rate) / 1000000LL); + +unlock_and_fail: + + pa_threaded_mainloop_unlock(pulsesrc->mainloop); + return 0; +} + +static GstStateChangeReturn gst_pulsesrc_change_state(GstElement *element, GstStateChange transition) { + GstPolypSrc *this = GST_PULSESRC(element); + + switch (transition) { + case GST_STATE_CHANGE_NULL_TO_READY: + + if (!this->mixer) + this->mixer = gst_pulsemixer_ctrl_new(this->server, this->device, GST_PULSEMIXER_SOURCE); + + break; + + case GST_STATE_CHANGE_READY_TO_NULL: + + if (this->mixer) { + gst_pulsemixer_ctrl_free(this->mixer); + this->mixer = NULL; + } + + break; + + default: + ; + } + + if (GST_ELEMENT_CLASS(parent_class)->change_state) + return GST_ELEMENT_CLASS(parent_class)->change_state(element, transition); + + return GST_STATE_CHANGE_SUCCESS; +} + +GType gst_pulsesrc_get_type(void) { + static GType pulsesrc_type = 0; + + if (!pulsesrc_type) { + + static const GTypeInfo pulsesrc_info = { + sizeof(GstPolypSrcClass), + gst_pulsesrc_base_init, + NULL, + gst_pulsesrc_class_init, + NULL, + NULL, + sizeof(GstPolypSrc), + 0, + gst_pulsesrc_init, + }; + + pulsesrc_type = g_type_register_static( + GST_TYPE_AUDIO_SRC, + "GstPolypSrc", + &pulsesrc_info, + 0); + + gst_pulsesrc_init_interfaces(pulsesrc_type); + } + + return pulsesrc_type; +} |