Re #1570: Update to work with audio routing (speaker vs handset). Thanks Bob Cripps for the patch!
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@4340 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/pjmedia/src/pjmedia-audiodev/bb10_dev.c b/pjmedia/src/pjmedia-audiodev/bb10_dev.c
index b963d00..a34707f 100644
--- a/pjmedia/src/pjmedia-audiodev/bb10_dev.c
+++ b/pjmedia/src/pjmedia-audiodev/bb10_dev.c
@@ -122,6 +122,7 @@
int quit;
/* Playback */
+ unsigned int pb_ctrl_audio_manager_handle;
snd_pcm_t *pb_pcm;
unsigned int pb_audio_manager_handle;
unsigned long pb_frames; /* samples_per_frame */
@@ -168,8 +169,6 @@
{
pjmedia_aud_dev_info *adi;
int pb_result, ca_result;
- int card = -1;
- int dev = 0;
unsigned int handle;
snd_pcm_t *pcm_handle;
@@ -183,17 +182,12 @@
if ((pb_result = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT,
&pcm_handle,
&handle,
- "/dev/snd/voicep",
+ (char*)"voice",
SND_PCM_OPEN_PLAYBACK))
>= 0)
{
- if ((pb_result = snd_pcm_plugin_set_disable (pcm_handle, PLUGIN_DISABLE_MMAP)) < 0) {
- TRACE_((THIS_FILE, "snd_pcm_plugin_set_disable ret = %d", pb_result));
- }else{
- TRACE_((THIS_FILE, "Try to open the device for playback - success"));
- }
- snd_pcm_close (pcm_handle);
- audio_manager_free_handle(handle);
+ snd_pcm_close (pcm_handle);
+ audio_manager_free_handle(handle);
} else {
TRACE_((THIS_FILE, "Try to open the device for playback - failure"));
}
@@ -201,17 +195,12 @@
if ((ca_result = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT,
&pcm_handle,
&handle,
- "/dev/snd/voicec",
+ (char*)"voice",
SND_PCM_OPEN_CAPTURE))
>= 0)
{
- if ((ca_result = snd_pcm_plugin_set_disable (pcm_handle, PLUGIN_DISABLE_MMAP)) < 0) {
- TRACE_((THIS_FILE, "snd_pcm_plugin_set_disable ret = %d", ca_result));
- }else{
- TRACE_((THIS_FILE, "Try to open the device for capture - success"));
- }
- snd_pcm_close (pcm_handle);
- audio_manager_free_handle(handle);
+ snd_pcm_close (pcm_handle);
+ audio_manager_free_handle(handle);
} else {
TRACE_((THIS_FILE, "Try to open the device for capture - failure"));
@@ -396,10 +385,16 @@
if (stream != NULL && stream->pb_pcm != NULL) {
snd_pcm_close(stream->pb_pcm);
stream->pb_pcm = NULL;
- if(stream->pb_audio_manager_handle != 0){
+
+ if (stream->pb_audio_manager_handle != 0) {
audio_manager_free_handle(stream->pb_audio_manager_handle);
stream->pb_audio_manager_handle = 0;
}
+
+ if (stream->pb_ctrl_audio_manager_handle != 0) {
+ audio_manager_free_handle(stream->pb_ctrl_audio_manager_handle);
+ stream->pb_ctrl_audio_manager_handle = 0;
+ }
}
}
@@ -415,7 +410,8 @@
if (stream != NULL && stream->ca_pcm != NULL) {
snd_pcm_close(stream->ca_pcm);
stream->ca_pcm = NULL;
- if(stream->ca_audio_manager_handle != 0){
+
+ if (stream->ca_audio_manager_handle != 0) {
audio_manager_free_handle(stream->ca_audio_manager_handle);
stream->ca_audio_manager_handle = 0;
}
@@ -457,7 +453,7 @@
{
close_play_pcm(stream);
TRACE_((THIS_FILE, "pb_thread_func failed prepare = %d", result));
- return PJ_SUCCESS;
+ return PJ_SUCCESS;
}
while (!stream->quit) {
@@ -470,6 +466,7 @@
frame.timestamp.u64 = tstamp.u64;
frame.bit_info = 0;
+ /* Read the audio from pjmedia */
result = stream->pb_cb (user_data, &frame);
if (result != PJ_SUCCESS || stream->quit)
break;
@@ -480,28 +477,37 @@
/* Write 640 to play unit */
result = snd_pcm_plugin_write(stream->pb_pcm,buf,size);
if (result != size || result < 0) {
- snd_pcm_channel_status_t status;
+ /* either the write to output device has failed or not the
+ * full amount of bytes have been written. This usually happens
+ * when audio routing is being changed by another thread
+ * Use a status variable for reading the error
+ */
+ snd_pcm_channel_status_t status;
+ status.channel = SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_plugin_status (stream->pb_pcm, &status) < 0) {
- PJ_LOG(4,(THIS_FILE,
+ /* Call has failed nothing we can do except log and
+ * continue */
+ PJ_LOG(4,(THIS_FILE,
"underrun: playback channel status error"));
- continue;
- }
-
- if (status.status == SND_PCM_STATUS_READY ||
- status.status == SND_PCM_STATUS_UNDERRUN)
- {
- if (snd_pcm_plugin_prepare (stream->pb_pcm,
- SND_PCM_CHANNEL_PLAYBACK) < 0)
- {
- PJ_LOG(4,(THIS_FILE,
- "underrun: playback channel prepare error"));
- continue;
- }
- }
-
- TRACE_((THIS_FILE, "pb_thread_func failed write = %d", result));
+ } else {
+ /* The status of the error has been read
+ * RIM say these are expected so we can "re-prepare" the stream
+ */
+ PJ_LOG(4,(THIS_FILE,"PLAY thread ERROR status = %d",
+ status.status));
+ if (status.status == SND_PCM_STATUS_READY ||
+ status.status == SND_PCM_STATUS_UNDERRUN ||
+ status.status == SND_PCM_STATUS_ERROR )
+ {
+ if (snd_pcm_plugin_prepare (stream->pb_pcm,
+ SND_PCM_CHANNEL_PLAYBACK) < 0)
+ {
+ PJ_LOG(4,(THIS_FILE,
+ "underrun: playback channel prepare error"));
+ }
+ }
+ }
}
-
tstamp.u64 += nframes;
}
@@ -552,7 +558,7 @@
{
close_capture_pcm(stream);
TRACE_((THIS_FILE, "ca_thread_func failed prepare = %d", result));
- return PJ_SUCCESS;
+ return PJ_SUCCESS;
}
while (!stream->quit) {
@@ -560,30 +566,48 @@
pj_bzero (buf, size);
+ /* read the input device */
result = snd_pcm_plugin_read(stream->ca_pcm, buf,size);
if(result <0 || result != size) {
+ /* We expect result to be size (640)
+ * It's not so we have to read the status error and "prepare"
+ * the channel. This usually happens when output audio routing
+ * has been changed by another thread.
+ * We won't "continue", instead just do what we can and leave
+ * the end of the loop to write what's in the buffer. Not entirely
+ * correct but saves a potential underrun in PJMEDIA
+ */
+ PJ_LOG (4,(THIS_FILE,
+ "snd_pcm_plugin_read ERROR read = %d required = %d",
+ result,size));
snd_pcm_channel_status_t status;
- if (snd_pcm_plugin_status (stream->ca_pcm, &status) < 0) {
- PJ_LOG (4,(THIS_FILE, "overrun: capture channel status "
- "error"));
- continue;
- }
-
- if (status.status == SND_PCM_STATUS_READY ||
- status.status == SND_PCM_STATUS_OVERRUN) {
- if (snd_pcm_plugin_prepare (stream->ca_pcm,
- SND_PCM_CHANNEL_CAPTURE) < 0)
- {
- PJ_LOG (4,(THIS_FILE, "overrun: capture channel prepare "
- "error"));
- continue;
- }
+ status.channel = SND_PCM_CHANNEL_CAPTURE;
+ if ((result = snd_pcm_plugin_status (stream->ca_pcm, &status)) < 0)
+ {
+ /* Should not fail but all we can do is continue */
+ PJ_LOG(4,(THIS_FILE, "capture: snd_pcm_plugin_status ret = %d",
+ result));
+ } else {
+ /* RIM say these are the errors that we should "prepare"
+ * after */
+ if (status.status == SND_PCM_STATUS_READY ||
+ status.status == SND_PCM_STATUS_OVERRUN ||
+ status.status == SND_PCM_STATUS_ERROR)
+ {
+ if (snd_pcm_plugin_prepare (stream->ca_pcm,
+ SND_PCM_CHANNEL_CAPTURE) < 0)
+ {
+ PJ_LOG (4,(THIS_FILE,
+ "overrun: capture channel prepare error"));
+ }
+ }
}
}
if (stream->quit)
break;
+ /* Write the capture audio data to PJMEDIA */
frame.type = PJMEDIA_FRAME_TYPE_AUDIO;
frame.buf = (void *) buf;
frame.size = size;
@@ -604,12 +628,62 @@
return PJ_SUCCESS;
}
+/* Audio routing, speaker/headset */
+static pj_status_t bb10_initialize_playback_ctrl(struct bb10_stream *stream,
+ bool speaker)
+{
+ /* Although the play and capture have audio manager handles, audio routing
+ * requires a separate handle
+ */
+ int ret = PJ_SUCCESS;
+
+ if (stream->pb_ctrl_audio_manager_handle == 0) {
+ /* lazy init an audio manager handle */
+ ret = audio_manager_get_handle(AUDIO_TYPE_VIDEO_CHAT, 0, false,
+ &stream->pb_ctrl_audio_manager_handle);
+ if (ret != 0) {
+ TRACE_((THIS_FILE, "audio_manager_get_handle ret = %d",ret));
+ return PJMEDIA_EAUD_SYSERR;
+ }
+ }
+
+ /* Set for either speaker or earpiece */
+ if (speaker) {
+ ret = audio_manager_set_handle_type(
+ stream->pb_ctrl_audio_manager_handle,
+ AUDIO_TYPE_VIDEO_CHAT,
+ AUDIO_DEVICE_SPEAKER,
+ AUDIO_DEVICE_DEFAULT);
+ } else {
+ ret = audio_manager_set_handle_type(
+ stream->pb_ctrl_audio_manager_handle,
+ AUDIO_TYPE_VIDEO_CHAT,
+ AUDIO_DEVICE_HANDSET,
+ AUDIO_DEVICE_DEFAULT);
+ }
+
+ if (ret == 0) {
+ /* RIM recommend this call */
+ ret = audio_manager_set_handle_routing_conditions(
+ stream->pb_ctrl_audio_manager_handle,
+ SETTINGS_RESET_ON_DEVICE_CONNECTION);
+ if (ret != 0) {
+ TRACE_((THIS_FILE,
+ "audio_manager_set_handle_routing_conditions ret = %d",
+ ret));
+ return PJMEDIA_EAUD_SYSERR;
+ }
+ } else {
+ TRACE_((THIS_FILE, "audio_manager_set_handle_type ret = %d", ret));
+ return PJMEDIA_EAUD_SYSERR;
+ }
+
+ return PJ_SUCCESS;
+}
static pj_status_t bb10_open_playback (struct bb10_stream *stream,
const pjmedia_aud_param *param)
{
- int card = -1;
- int dev = 0;
int ret = 0;
snd_pcm_channel_info_t pi;
snd_pcm_channel_setup_t setup;
@@ -622,51 +696,54 @@
return PJMEDIA_EAUD_INVDEV;
}
- if ((ret = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT,
- &stream->pb_pcm, &stream->pb_audio_manager_handle,
- "/dev/snd/voicep",
- SND_PCM_OPEN_PLAYBACK)) < 0)
+ /* Use the bb10 audio manager API to open as opposed to QNX core audio
+ * Echo cancellation built in
+ */
+ if ((ret = audio_manager_snd_pcm_open_name(
+ AUDIO_TYPE_VIDEO_CHAT,
+ &stream->pb_pcm, &stream->pb_audio_manager_handle,
+ (char*)"voice",
+ SND_PCM_OPEN_PLAYBACK)) < 0)
{
TRACE_((THIS_FILE, "audio_manager_snd_pcm_open_name ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
- ret = audio_manager_set_handle_type(stream->pb_audio_manager_handle, AUDIO_TYPE_VIDEO_CHAT, AUDIO_DEVICE_HANDSET , AUDIO_DEVICE_HANDSET);
- if (ret==0) {
- ret = audio_manager_set_handle_routing_conditions(stream->pb_audio_manager_handle, SETTINGS_RESET_ON_DEVICE_CONNECTION);
- if(ret != 0){
- TRACE_((THIS_FILE, "audio_manager_set_handle_routing_conditions ret = %d", ret));
- return PJMEDIA_EAUD_SYSERR;
- }
- }else{
- TRACE_((THIS_FILE, "audio_manager_set_handle_type ret = %d", ret));
- return PJMEDIA_EAUD_SYSERR;
+
+ /* Required call from January 2013 gold OS release */
+ if ((ret = snd_pcm_plugin_set_disable(stream->pb_pcm,
+ PLUGIN_DISABLE_MMAP)) < 0)
+ {
+ TRACE_((THIS_FILE, "snd_pcm_plugin_set_disable ret = %d", ret));
+ return PJMEDIA_EAUD_SYSERR;
}
- if ((ret = snd_pcm_plugin_set_disable (stream->pb_pcm, PLUGIN_DISABLE_MMAP)) < 0) {
- TRACE_((THIS_FILE, "snd_pcm_plugin_set_disable ret = %d", ret));
- return PJMEDIA_EAUD_SYSERR;
+ /* Required call from January 2013 gold OS release */
+ if ((ret = snd_pcm_plugin_set_enable(stream->pb_pcm,
+ PLUGIN_ROUTING)) < 0)
+ {
+ TRACE_((THIS_FILE, "snd_pcm_plugin_set_enable ret = %d", ret));
+ return PJMEDIA_EAUD_SYSERR;
}
+
/* TODO PJ_ZERO */
memset (&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((ret = snd_pcm_plugin_info (stream->pb_pcm, &pi)) < 0) {
- TRACE_((THIS_FILE, "snd_pcm_plugin_info ret = %d", ret));
- return PJMEDIA_EAUD_SYSERR;
+ TRACE_((THIS_FILE, "snd_pcm_plugin_info ret = %d", ret));
+ return PJMEDIA_EAUD_SYSERR;
}
memset (&pp, 0, sizeof (pp));
- /* Request VoIP compatible capabilities
- * On simulator frag_size is always negotiated to 170
- */
+ /* Request VoIP compatible capabilities */
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_PLAYBACK;
pp.start_mode = SND_PCM_START_DATA;
pp.stop_mode = SND_PCM_STOP_ROLLOVER;
/* HARD CODE for the time being PJMEDIA expects 640 for 16khz */
pp.buf.block.frag_size = PREFERRED_FRAME_SIZE*2;
- /* Increasing this internal buffer count delays write failure in the loop */
- pp.buf.block.frags_max = 4;
+ /* RIM recommends maximum of 3 */
+ pp.buf.block.frags_max = 3;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
/* HARD CODE for the time being PJMEDIA expects 16khz */
@@ -714,7 +791,7 @@
TRACE_((THIS_FILE, "bb10_open_playback: pb_frames = %d clock = %d",
stream->pb_frames, param->clock_rate));
-
+
return PJ_SUCCESS;
}
@@ -724,8 +801,6 @@
int ret = 0;
unsigned int rate;
unsigned long tmp_buf_size;
- int card = -1;
- int dev = 0;
int frame_size;
snd_pcm_channel_info_t pi;
snd_mixer_group_t group;
@@ -735,19 +810,29 @@
if (param->rec_id < 0 || param->rec_id >= stream->af->dev_cnt)
return PJMEDIA_EAUD_INVDEV;
- if ((ret = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT,
- &stream->ca_pcm,
- &stream->ca_audio_manager_handle,
- "/dev/snd/voicec",
- SND_PCM_OPEN_CAPTURE)) < 0)
+ if ((ret=audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT,
+ &stream->ca_pcm,
+ &stream->ca_audio_manager_handle,
+ (char*)"voice",
+ SND_PCM_OPEN_CAPTURE)) < 0)
{
TRACE_((THIS_FILE, "audio_manager_snd_pcm_open_name ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
- if ((ret = snd_pcm_plugin_set_disable (stream->ca_pcm, PLUGIN_DISABLE_MMAP)) < 0) {
+ /* Required call from January 2013 gold OS release */
+ if ((ret = snd_pcm_plugin_set_disable (stream->ca_pcm,
+ PLUGIN_DISABLE_MMAP)) < 0)
+ {
TRACE_(("snd_pcm_plugin_set_disable failed: %d",ret));
return PJMEDIA_EAUD_SYSERR;
}
+ /* Required call from January 2013 gold OS release */
+ if ((ret = snd_pcm_plugin_set_enable(stream->ca_pcm,
+ PLUGIN_ROUTING)) < 0)
+ {
+ TRACE_(("snd_pcm_plugin_set_enable failed: %d",ret));
+ return PJMEDIA_EAUD_SYSERR;
+ }
/* sample reads the capabilities of the capture */
memset (&pi, 0, sizeof (pi));
@@ -769,8 +854,8 @@
pp.stop_mode = SND_PCM_STOP_ROLLOVER;
/* HARD CODE for the time being PJMEDIA expects 640 for 16khz */
pp.buf.block.frag_size = PREFERRED_FRAME_SIZE*2;
- /* Not applicable for capture hence -1 */
- pp.buf.block.frags_max = -1;
+ /* From January 2013 gold OS release. RIM recommend these for capture */
+ pp.buf.block.frags_max = 1;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
/* HARD CODE for the time being PJMEDIA expects 16khz */
@@ -802,11 +887,8 @@
} else {
}
- /* frag_size should be 160 */
frame_size = setup.buf.block.frag_size;
- /* END BB10 init */
-
/* Set clock rate */
rate = param->clock_rate;
stream->ca_frames = (unsigned long) param->samples_per_frame /
@@ -882,12 +964,24 @@
}
}
+ /* Part of the play functionality but the RIM/Truphone loopback sample
+ * initialializes after the play and capture
+ * "false" is default/earpiece for output
+ */
+ status = bb10_initialize_playback_ctrl(stream,false);
+ if (status != PJ_SUCCESS) {
+ return PJMEDIA_EAUD_SYSERR;
+ }
+
*p_strm = &stream->base;
return PJ_SUCCESS;
}
-/* API: get running parameter */
+/*
+ * API: get running parameter
+ * based on ALSA template
+ */
static pj_status_t bb10_stream_get_param(pjmedia_aud_stream *s,
pjmedia_aud_param *pi)
{
@@ -901,7 +995,10 @@
}
-/* API: get capability */
+/*
+ * API: get capability
+ * based on ALSA template
+*/
static pj_status_t bb10_stream_get_cap(pjmedia_aud_stream *s,
pjmedia_aud_dev_cap cap,
void *pval)
@@ -916,28 +1013,51 @@
/* Recording latency */
*(unsigned*)pval = stream->param.input_latency_ms;
return PJ_SUCCESS;
+
} else if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY &&
(stream->param.dir & PJMEDIA_DIR_PLAYBACK))
{
/* Playback latency */
*(unsigned*)pval = stream->param.output_latency_ms;
return PJ_SUCCESS;
+
} else {
return PJMEDIA_EAUD_INVCAP;
}
}
-/* API: set capability */
+/*
+ * API: set capability
+ * Currently just supporting toggle between speaker and earpiece
+ */
static pj_status_t bb10_stream_set_cap(pjmedia_aud_stream *strm,
pjmedia_aud_dev_cap cap,
const void *value)
{
- PJ_UNUSED_ARG(strm);
- PJ_UNUSED_ARG(cap);
- PJ_UNUSED_ARG(value);
+ pj_status_t ret = PJ_SUCCESS;
+ struct bb10_stream *stream = (struct bb10_stream*)strm;
- return PJMEDIA_EAUD_INVCAP;
+ if (cap != PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE || value == NULL) {
+ TRACE_((THIS_FILE,"bb10_stream_set_cap() = PJMEDIA_EAUD_INVCAP"));
+ return PJMEDIA_EAUD_INVCAP;
+
+ } else {
+ pjmedia_aud_dev_route route = *((pjmedia_aud_dev_route*)value);
+ /* Use the initialization function which lazy-inits the
+ * handle for routing
+ */
+ if (route == PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER) {
+ ret = bb10_initialize_playback_ctrl(stream,true);
+ } else {
+ ret = bb10_initialize_playback_ctrl(stream,false);
+ }
+ }
+
+ if (ret != PJ_SUCCESS) {
+ TRACE_((THIS_FILE,"bb10_stream_set_cap() = %d",ret));
+ }
+ return ret;
}