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Audio works in test

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This commit is contained in:
mannol 2015-03-23 22:22:17 +01:00
parent 62af82705a
commit 995bddbc26
4 changed files with 19 additions and 6 deletions
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3
toxav/av_test.c
View File

@ -156,7 +156,7 @@ void iterate(Tox* Bsn, ToxAV* AliceAV, ToxAV* BobAV)
toxav_iteration(AliceAV);
toxav_iteration(BobAV);
c_sleep(20);
c_sleep(toxav_iteration_interval(AliceAV));
}
int device_read_frame(ALCdevice* device, int32_t frame_dur, int16_t* PCM, size_t max_size)
@ -652,7 +652,6 @@ int main (int argc, char** argv)
}
iterate(Bsn, AliceAV, BobAV);
// c_sleep(20);
}
{

19
toxav/codec.c
View File

@ -258,7 +258,7 @@ void cs_do(CSSession *cs)
if (cs->audio_decoder) { /* If receiving enabled */
RTPMessage *msg;
uint16_t fsize = 16000; /* Max frame size for 48 kHz */
uint16_t fsize = 10000; /* Should be enough for all normal frequences */
int16_t tmp[fsize * 2];
while ((msg = jbuf_read(cs->j_buf, &success)) || success == 2) {
@ -293,6 +293,7 @@ void cs_do(CSSession *cs)
LOGGER_WARNING("Decoding error: %s", opus_strerror(rc));
} else if (cs->acb.first) {
/* Play */
LOGGER_DEBUG("Playing audio frame size: %d chans: %d srate: %d", rc, cs->last_pack_channels, cs->last_packet_sampling_rate);
cs->acb.first(cs->agent, cs->friend_id, tmp, rc,
cs->last_pack_channels, cs->last_packet_sampling_rate, cs->acb.second);
}
@ -598,6 +599,7 @@ int cs_set_sending_audio_bitrate(CSSession *cs, int32_t rate)
return -1;
}
LOGGER_DEBUG("Set new encoder bitrate to: %d", rate);
return 0;
}
@ -607,6 +609,9 @@ int cs_set_sending_audio_sampling_rate(CSSession* cs, int32_t rate)
if (cs->audio_encoder == NULL)
return -1;
if (cs->encoder_sample_rate == rate)
return 0;
int rc = OPUS_OK;
int bitrate = 0;
int channels = cs->encoder_channels;
@ -619,6 +624,9 @@ int cs_set_sending_audio_sampling_rate(CSSession* cs, int32_t rate)
}
cs_disable_audio_sending(cs);
cs->encoder_sample_rate = rate;
LOGGER_DEBUG("Set new encoder sampling rate: %d", rate);
return cs_enable_audio_sending(cs, bitrate, channels);
}
@ -642,6 +650,8 @@ int cs_set_sending_audio_channels(CSSession* cs, int32_t count)
}
cs_disable_audio_sending(cs);
LOGGER_DEBUG("Set new encoder channel count: %d", count);
return cs_enable_audio_sending(cs, bitrate, count);
}
@ -674,8 +684,12 @@ int cs_enable_audio_sending(CSSession* cs, uint32_t bitrate, int channels)
if (cs->audio_encoder)
return 0;
if (!cs->encoder_sample_rate)
cs->encoder_sample_rate = 48000;
cs->encoder_channels = channels;
int rc = OPUS_OK;
cs->audio_encoder = opus_encoder_create(48000, channels, OPUS_APPLICATION_AUDIO, &rc);
cs->audio_encoder = opus_encoder_create(cs->encoder_sample_rate, channels, OPUS_APPLICATION_AUDIO, &rc);
if ( rc != OPUS_OK ) {
LOGGER_ERROR("Error while starting audio encoder: %s", opus_strerror(rc));
@ -696,7 +710,6 @@ int cs_enable_audio_sending(CSSession* cs, uint32_t bitrate, int channels)
goto FAILURE;
}
cs->encoder_channels = channels;
return 0;
FAILURE:

1
toxav/codec.h
View File

@ -108,6 +108,7 @@ typedef struct CSSession_s {
/* audio encoding */
OpusEncoder *audio_encoder;
int32_t encoder_channels;
int32_t encoder_sample_rate;
/* audio decoding */
OpusDecoder *audio_decoder;

2
toxav/toxav.c
View File

@ -957,7 +957,7 @@ bool call_prepare_transmission(ToxAVCall* call)
call->rtps[audio_index]->cs = call->cs;
/* Only enable sending if bitrate is defined */
if (call->s_audio_b > 0 && cs_enable_audio_sending(call->cs, call->s_audio_b, 2) != 0) {
if (call->s_audio_b > 0 && cs_enable_audio_sending(call->cs, call->s_audio_b * 1000, 2) != 0) {
LOGGER_WARNING("Failed to enable audio sending!");
goto FAILURE;
}