ims-volte-vowifi
/
doubango
9
0
Fork 0
Code Issues Pull Requests Projects Releases Wiki Activity

Fix crash on session update (audio <-> video)

This commit is contained in:
Doubango Telecom 2017-02-09 23:09:51 +01:00
parent ff03a51e4c
commit 1e5567d55b
1 changed files with 20 additions and 17 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

37
tinyDAV/src/audio/tdav_session_audio.c
View File

@ -175,6 +175,7 @@ static int tdav_session_audio_producer_enc_cb(const void* callback_data, const v
tdav_session_audio_t* audio = (tdav_session_audio_t*)callback_data;
tdav_session_av_t* base = (tdav_session_av_t*)callback_data;
struct tmedia_codec_s* codec_copy = tsk_null; // to ensure thread-safeness (crash reported by Chunghwa Telecom), SIP reINVITE or UPDATE could update the encoder's codec
if (!audio) {
TSK_DEBUG_ERROR("Null session");
@ -215,32 +216,39 @@ static int tdav_session_audio_producer_enc_cb(const void* callback_data, const v
}
tsk_safeobj_unlock(base);
}
// get a copy
if (!(codec_copy = tsk_object_ref(audio->encoder.codec))) {
TSK_DEBUG_ERROR("Null encoder");
goto done;
}
// check if we're sending DTMF or not
if (audio->is_sending_dtmf_events) {
if (base->rtp_manager) {
// increment the timestamp
base->rtp_manager->rtp.timestamp += TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec)/*duration*/;
base->rtp_manager->rtp.timestamp += TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(codec_copy)/*duration*/;
}
TSK_DEBUG_INFO("Skiping audio frame as we're sending DTMF...");
return 0;
goto done;
}
// resample if needed
if (base->producer->audio.rate != audio->encoder.codec->out.rate || base->producer->audio.channels != TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels) {
if (base->producer->audio.rate != codec_copy->out.rate || base->producer->audio.channels != TMEDIA_CODEC_AUDIO(codec_copy)->out.channels) {
tsk_size_t resampler_result_size = 0;
int bytesPerSample = (base->producer->audio.bits_per_sample >> 3);
if (!audio->encoder.resampler.instance) {
TSK_DEBUG_INFO("Create audio resampler(%s) for producer: rate=%d->%d, channels=%d->%d, bytesPerSample=%d",
audio->encoder.codec->plugin->desc,
base->producer->audio.rate, audio->encoder.codec->out.rate,
base->producer->audio.channels, TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels,
codec_copy->plugin->desc,
base->producer->audio.rate, codec_copy->out.rate,
base->producer->audio.channels, TMEDIA_CODEC_AUDIO(codec_copy)->out.channels,
bytesPerSample);
audio->encoder.resampler.instance = _tdav_session_audio_resampler_create(
bytesPerSample,
base->producer->audio.rate, audio->encoder.codec->out.rate,
base->producer->audio.rate, codec_copy->out.rate,
base->producer->audio.ptime,
base->producer->audio.channels, TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels,
base->producer->audio.channels, TMEDIA_CODEC_AUDIO(codec_copy)->out.channels,
TDAV_AUDIO_RESAMPLER_DEFAULT_QUALITY,
&audio->encoder.resampler.buffer, &audio->encoder.resampler.buffer_size
);
@ -275,19 +283,14 @@ static int tdav_session_audio_producer_enc_cb(const void* callback_data, const v
}
// Encode data
if ((audio->encoder.codec = tsk_object_ref(audio->encoder.codec))) { /* Thread safeness (SIP reINVITE or UPDATE could update the encoder) */
out_size = audio->encoder.codec->plugin->encode(audio->encoder.codec, buffer, size, &audio->encoder.buffer, &audio->encoder.buffer_size);
if (out_size) {
trtp_manager_send_rtp(base->rtp_manager, audio->encoder.buffer, out_size, TMEDIA_CODEC_FRAME_DURATION_AUDIO_ENCODING(audio->encoder.codec), tsk_false/*Marker*/, tsk_true/*lastPacket*/);
}
tsk_object_unref(audio->encoder.codec);
}
else {
TSK_DEBUG_WARN("No encoder");
out_size = codec_copy->plugin->encode(codec_copy, buffer, size, &audio->encoder.buffer, &audio->encoder.buffer_size);
if (out_size) {
trtp_manager_send_rtp(base->rtp_manager, audio->encoder.buffer, out_size, TMEDIA_CODEC_FRAME_DURATION_AUDIO_ENCODING(codec_copy), tsk_false/*Marker*/, tsk_true/*lastPacket*/);
}
}
done:
tsk_object_unref(codec_copy);
return ret;
}