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PortAudio doesn't make sure that a default device is even available on the default Host API (go figure). 

Therefore opening the default stream may fail in the precense of usable devices, on other Host API's.
If the default stream fails to open iterate among the Host API's to find one with a default device.

svn path=/trunk/; revision=31318
This commit is contained in:
Jaap Keuter 2009-12-18 23:26:12 +00:00
parent 7d11c07ace
commit 220054bd43
1 changed files with 130 additions and 40 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

170
gtk/rtp_player.c
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@ -28,12 +28,12 @@
* - The VoipCalls will call add_rtp_packet() every time there is an RTP
* packet
* - add_rtp_packet() will add the RTP packet in a RTP stream struct, and
* create the RTP stream if it is the first RTP in the stream.
* create the RTP stream if it is the first RTP packet in the stream.
* - Each new RTP stream will be added to a list of RTP streams, called
* rtp_streams_list
* - When the user clicks "Player" in the VoipCall dialogue,
* rtp_player_init() is called.
* - rtp_player_init() create the main dialog, and it calls:
* - rtp_player_init() creates the main dialog, and it calls:
* + mark_rtp_stream_to_play() to mark the RTP streams that needs to be
* displayed. These are the RTP streams that match the selected calls in
* the VoipCall dlg.
@ -46,7 +46,7 @@
* + add_channel_to_window() will create and add the Audio graphic
* representation in the main window
* - When the user clicks the check box to listen one of the Audio channels,
* the structure rtp_channels is filled to play one or two RTP channels
* the structure rtp_channels is filled to play one or two RTP channels
* (a max of two channels can be listened at a given moment)
*/
@ -131,7 +131,7 @@ static int new_jitter_buff;
/* a hash table with the RTP streams to play per audio channel */
static GHashTable *rtp_channels_hash = NULL;
/* Port Audio staff */
/* Port Audio stuff */
#define SAMPLE_RATE (8000)
#define NUM_CHANNELS (2)
@ -236,7 +236,7 @@ typedef struct _rtp_play_channles {
#endif /* PORTAUDIO_API_1 */
} rtp_play_channels_t;
/* The two RTP channles to play */
/* The two RTP channels to play */
static rtp_play_channels_t *rtp_channels = NULL;
typedef struct _rtp_decoder_t {
@ -370,7 +370,7 @@ add_rtp_packet(const struct _rtp_info *rtp_info, packet_info *pinfo)
pinfo->srcport, get_addr_name(&(pinfo->dst)),
pinfo->destport, rtp_info->info_sync_src );
/* lookup for this rtp packet in the stream hash table*/
/* lookup for this RTP packet in the stream hash table */
stream_info = g_hash_table_lookup( rtp_streams_hash, key_str->str);
/* if it is not in the hash table, create a new stream */
@ -394,7 +394,7 @@ add_rtp_packet(const struct _rtp_info *rtp_info, packet_info *pinfo)
rtp_streams_list = g_list_append(rtp_streams_list, stream_info);
}
/* increment the number of packets in this stream, this is used for the progress bar and statistics*/
/* increment the number of packets in this stream, this is used for the progress bar and statistics */
stream_info->num_packets++;
/* Add the RTP packet to the list */
@ -429,11 +429,11 @@ mark_rtp_stream_to_play(gchar *key _U_ , rtp_stream_info_t *rsi, gpointer ptr _U
voip_calls_info_t *tmp_voip_call;
/* Reset the "to be play" value because the user can close and reopen the RTP Player window
* and the streams are nor reset in that case
* and the streams are not reset in that case
*/
rsi->play = FALSE;
/* and associate the RTP stream with a call using the first RTP in the stream*/
/* and associate the RTP stream with a call using the first RTP packet in the stream */
graph_list = g_list_first(voip_calls->graph_analysis->list);
while (graph_list)
{
@ -784,12 +784,12 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr _U_)
silence.status = status;
g_array_append_val(rci->samples, silence);
/* only mark the fisrt in the silence that has the previos problem (S_DROP_BY_JITT or S_WRONG_SEQ ) */
/* only mark the first in the silence that has the previous problem (S_DROP_BY_JITT or S_WRONG_SEQ) */
status = S_NORMAL;
}
decoded_bytes_prev = 0;
start_timestamp = rp->info->info_timestamp; /* defined start_timestmp to avoid overflow in timestamp. TODO: handle the timestamp correctly */
start_timestamp = rp->info->info_timestamp; /* defined start_timestamp to avoid overflow in timestamp. TODO: handle the timestamp correctly */
start_rtp_time = 0;
start_time = (double)rp->arrive_offset/1000;
rtp_time_prev = 0;
@ -805,7 +805,7 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr _U_)
silence.status = status;
g_array_append_val(rci->samples, silence);
/* only mark the fisrt in the silence that has the previos problem (S_DROP_BY_JITT or S_WRONG_SEQ ) */
/* only mark the first in the silence that has the previous problem (S_DROP_BY_JITT or S_WRONG_SEQ) */
status = S_NORMAL;
}
@ -980,15 +980,15 @@ draw_channel_cursor(rtp_channel_info_t *rci, guint32 start_index)
g_signal_handlers_disconnect_by_func(rci->h_scrollbar_adjustment, h_scrollbar_changed, rci);
/* Move the horizontal scroll bar */
/* if ( (rci->cursor_prev/MULT < (rci->h_scrollbar_adjustment->value+rci->h_scrollbar_adjustment->page_increment)) &&
#if 0
if ( (rci->cursor_prev/MULT < (rci->h_scrollbar_adjustment->value+rci->h_scrollbar_adjustment->page_increment)) &&
(index/MULT >= (rci->h_scrollbar_adjustment->value+rci->h_scrollbar_adjustment->page_increment)) ){
for (i=1; i<10; i++) {
rci->h_scrollbar_adjustment->value += rci->h_scrollbar_adjustment->page_size/10;
gtk_adjustment_value_changed(rci->h_scrollbar_adjustment);
}
}
*/
#endif
if (!rci->cursor_catch) {
if (index/MULT < rci->h_scrollbar_adjustment->page_size/2) {
rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->lower;
@ -1013,8 +1013,8 @@ draw_channel_cursor(rtp_channel_info_t *rci, guint32 start_index)
/* Connect back the "value" scroll signal */
g_signal_connect(rci->h_scrollbar_adjustment, "value_changed", G_CALLBACK(h_scrollbar_changed), rci);
/* if (index/MULT < rci->h_scrollbar_adjustment->page_increment) {
#if 0
if (index/MULT < rci->h_scrollbar_adjustment->page_increment) {
rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->lower;
} else if (index/MULT > (rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size + rci->h_scrollbar_adjustment->page_increment)) {
rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size;
@ -1023,18 +1023,21 @@ draw_channel_cursor(rtp_channel_info_t *rci, guint32 start_index)
rci->h_scrollbar_adjustment->value = index/MULT;
}
}
*/
#endif
/* if (index/MULT < rci->h_scrollbar_adjustment->page_size/2) {
#if 0
if (index/MULT < rci->h_scrollbar_adjustment->page_size/2) {
rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->lower;
} else if (index/MULT > (rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size/2)) {
rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size;
} else {
rci->h_scrollbar_adjustment->value = index/MULT - rci->h_scrollbar_adjustment->page_size/2;
}
*/
/* gtk_adjustment_value_changed(rci->h_scrollbar_adjustment);
*/
#endif
#if 0
gtk_adjustment_value_changed(rci->h_scrollbar_adjustment);
#endif
rci->cursor_prev = index;
}
@ -1380,7 +1383,7 @@ configure_event_channels(GtkWidget *widget, GdkEventConfigure *event _U_)
rtp_channel_info_t *rci;
int i;
/* the first calor is blue to highlight the selected item
/* the first color is blue to highlight the selected item
* the other collors are the same as in the Voip Graph analysys
* to match the same calls
*/
@ -1421,7 +1424,7 @@ configure_event_channels(GtkWidget *widget, GdkEventConfigure *event _U_)
widget->allocation.width,
widget->allocation.height);
/* create gcs for the background color of each channel */
/* create gc's for the background color of each channel */
for (i=0; i<MAX_NUM_COL_CONV+1; i++){
rci->bg_gc[i]=gdk_gc_new(rci->pixmap);
gdk_gc_set_rgb_fg_color(rci->bg_gc[i], &col[i]);
@ -1602,7 +1605,7 @@ play_channels(void)
exit(10);
}
/* if we are in PAUSE change the sate */
/* if we are in PAUSE change the state */
if (rtp_channels->pause) {
rtp_channels->pause = FALSE;
/* set the sensitive state of the buttons (decode, play, pause, stop) */
@ -1621,26 +1624,14 @@ play_channels(void)
NUM_CHANNELS, /* Stereo output */
PA_SAMPLE_TYPE, /* 16 bit Integer output */
NULL,
SAMPLE_RATE,
SAMPLE_RATE, /* 8 kHz */
FRAMES_PER_BUFFER,
0, /* number of buffers, if zero then use default minimum */
paClipOff, /* we won't output out of range samples so don't bother clipping them */
paCallback,
rtp_channels );
#else /* PORTAUDIO_API_1 */
err = Pa_OpenDefaultStream(
&pa_stream,
0,
NUM_CHANNELS,
PA_SAMPLE_TYPE,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paCallback,
rtp_channels );
#endif /* PORTAUDIO_API_1 */
if( err != paNoError ) {
#if PORTAUDIO_API_1
const char *deviceName = "No Device";
PaDeviceID device = Pa_GetDefaultOutputDeviceID();
@ -1660,7 +1651,75 @@ play_channels(void)
" Default deviceName: %s (%d)", deviceName, device);
gtk_dialog_run (GTK_DIALOG (dialog));
gtk_widget_destroy (dialog);
#else
dialog = gtk_message_dialog_new ((GtkWindow *) rtp_player_dlg_w,
GTK_DIALOG_DESTROY_WITH_PARENT, GTK_MESSAGE_ERROR,GTK_BUTTONS_CLOSE,
"Can not Open Stream in PortAudio Library.\n Error: %s", Pa_GetErrorText( err ));
gtk_dialog_run (GTK_DIALOG (dialog));
gtk_widget_destroy (dialog);
return;
}
#else /* PORTAUDIO_API_1 */
if (Pa_GetDefaultOutputDevice() != paNoDevice) {
err = Pa_OpenDefaultStream(
&pa_stream,
0,
NUM_CHANNELS, /* Stereo output */
PA_SAMPLE_TYPE, /* 16 bit Integer output */
SAMPLE_RATE, /* 8 kHz */
FRAMES_PER_BUFFER,
paCallback,
rtp_channels );
} else {
/* If the Default Host API doesn't even provide a device
* we might as well go look for another.
*/
PaHostApiIndex host_api_count = Pa_GetHostApiCount();
PaHostApiIndex default_host_api_index = Pa_GetDefaultHostApi();
PaHostApiIndex host_api_index;
const PaHostApiInfo *host_api_info;
for (host_api_index=0; host_api_index<host_api_count; host_api_index++)
{
/* Skip the default host API, that didn't work before */
if (host_api_index == default_host_api_index)
continue;
/* If we find a host API with a device, then take it. */
host_api_info = Pa_GetHostApiInfo(host_api_index);
if (host_api_info->deviceCount > 0)
break;
}
if (host_api_index<host_api_count)
{
PaStreamParameters stream_parameters;
stream_parameters.device = host_api_info->defaultOutputDevice;
stream_parameters.channelCount = NUM_CHANNELS; /* Stereo output */
stream_parameters.sampleFormat = PA_SAMPLE_TYPE; /* 16 bit Integer output */
stream_parameters.suggestedLatency = 0;
stream_parameters.hostApiSpecificStreamInfo = NULL;
#if DEBUG
g_print("Trying Host API: %s\n", host_api_info->name);
#endif
err = Pa_OpenStream(
&pa_stream,
NULL, /* no input */
&stream_parameters,
SAMPLE_RATE, /* 8 kHz */
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
paCallback,
rtp_channels );
}
else
{
err = paNoDevice;
}
}
if( err != paNoError ) {
PaHostApiIndex hostApi = Pa_GetDefaultHostApi();
if (hostApi < 0)
{
@ -1708,7 +1767,7 @@ play_channels(void)
gtk_widget_destroy (dialog);
}
}
#endif
dialog = gtk_message_dialog_new ((GtkWindow *) rtp_player_dlg_w,
GTK_DIALOG_DESTROY_WITH_PARENT, GTK_MESSAGE_ERROR,GTK_BUTTONS_CLOSE,
"Can not Open Stream in PortAudio Library.\n Error: %s", Pa_GetErrorText( err ));
@ -1716,6 +1775,7 @@ play_channels(void)
gtk_widget_destroy (dialog);
return;
}
#endif
err = Pa_StartStream( pa_stream );
if( err != paNoError ) {
@ -2142,6 +2202,36 @@ rtp_player_init(voip_calls_tapinfo_t *voip_calls_tap)
reset_rtp_channels();
#if DEBUG
g_print("Pa_GetHostApiCount() = %d\n", Pa_GetHostApiCount());
g_print("Pa_GetDefaultHostApi() = %d\n", Pa_GetDefaultHostApi());
if ((Pa_GetHostApiCount() >= 0) && (Pa_GetDefaultHostApi() >= 0))
{
unsigned int i;
PaHostApiIndex api_index;
const PaHostApiInfo *api_info = Pa_GetHostApiInfo( (unsigned int)Pa_GetDefaultHostApi() );
g_print("Default PaHostApiInfo.type = %d (%s)\n", api_info->type, api_info->name);
for (i=0; i<(unsigned int)Pa_GetHostApiCount(); i++)
{
api_info = Pa_GetHostApiInfo( i );
g_print("PaHostApiInfo[%u].type = %d (%s)\n", i, api_info->type, api_info->name);
}
api_index = Pa_HostApiTypeIdToHostApiIndex( paALSA );
if (api_index < 0)
{
g_print("api_index for paALSA not found (%d)\n", api_index);
}
else
{
api_info = Pa_GetHostApiInfo( (unsigned int)api_index );
g_print("This should be ALSA: %s\n", api_info->name);
}
}
#endif
/* create the dialog window */
rtp_player_dlg_create();