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Add some explanatory comments to the toxav audio code.

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By @zoff99.
This commit is contained in:
iphydf 2018-01-19 20:55:27 +00:00
parent 1a4e56397b
commit 656040027d
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GPG Key ID: 3855DBA2D74403C9
1 changed files with 51 additions and 0 deletions
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51
toxav/audio.c
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@ -180,6 +180,15 @@ void ac_iterate(ACSession *ac)
continue;
}
/*
* frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
* where
* packet is the byte array containing the compressed data
* len is the exact number of bytes contained in the packet
* decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
* max_size is the max duration of the frame in samples (per channel) that can fit
* into the decoded_frame array
*/
rc = opus_decode(ac->decoder, msg->data + 4, msg->len - 4, temp_audio_buffer, 5760, 0);
free(msg);
}
@ -356,6 +365,11 @@ static struct RTPMessage *jbuf_read(struct JitterBuffer *q, int32_t *success)
OpusEncoder *create_audio_encoder(Logger *log, int32_t bit_rate, int32_t sampling_rate, int32_t channel_count)
{
int status = OPUS_OK;
/*
* OPUS_APPLICATION_VOIP Process signal for improved speech intelligibility
* OPUS_APPLICATION_AUDIO Favor faithfulness to the original input
* OPUS_APPLICATION_RESTRICTED_LOWDELAY Configure the minimum possible coding delay
*/
OpusEncoder *rc = opus_encoder_create(sampling_rate, channel_count, OPUS_APPLICATION_VOIP, &status);
if (status != OPUS_OK) {
@ -363,6 +377,16 @@ OpusEncoder *create_audio_encoder(Logger *log, int32_t bit_rate, int32_t samplin
return NULL;
}
/*
* Rates from 500 to 512000 bits per second are meaningful as well as the special
* values OPUS_BITRATE_AUTO and OPUS_BITRATE_MAX. The value OPUS_BITRATE_MAX can
* be used to cause the codec to use as much rate as it can, which is useful for
* controlling the rate by adjusting the output buffer size.
*
* Parameters:
* [in] x opus_int32: bitrate in bits per second.
*/
status = opus_encoder_ctl(rc, OPUS_SET_BITRATE(bit_rate));
if (status != OPUS_OK) {
@ -370,6 +394,14 @@ OpusEncoder *create_audio_encoder(Logger *log, int32_t bit_rate, int32_t samplin
goto FAILURE;
}
/*
* Configures the encoder's use of inband forward error correction.
* Note:
* This is only applicable to the LPC layer
* Parameters:
* [in] x int: FEC flag, 0 (disabled) is default
*/
/* Enable in-band forward error correction in codec */
status = opus_encoder_ctl(rc, OPUS_SET_INBAND_FEC(1));
@ -378,6 +410,15 @@ OpusEncoder *create_audio_encoder(Logger *log, int32_t bit_rate, int32_t samplin
goto FAILURE;
}
/*
* Configures the encoder's expected packet loss percentage.
* Higher values with trigger progressively more loss resistant behavior in
* the encoder at the expense of quality at a given bitrate in the lossless case,
* but greater quality under loss.
* Parameters:
* [in] x int: Loss percentage in the range 0-100, inclusive.
*/
/* Make codec resistant to up to 10% packet loss
* NOTE This could also be adjusted on the fly, rather than hard-coded,
* with feedback from the receiving client.
@ -389,6 +430,16 @@ OpusEncoder *create_audio_encoder(Logger *log, int32_t bit_rate, int32_t samplin
goto FAILURE;
}
/*
* Configures the encoder's computational complexity.
*
* The supported range is 0-10 inclusive with 10 representing the highest complexity.
* The default value is 10.
*
* Parameters:
* [in] x int: 0-10, inclusive
*/
/* Set algorithm to the highest complexity, maximizing compression */
status = opus_encoder_ctl(rc, OPUS_SET_COMPLEXITY(AUDIO_OPUS_COMPLEXITY));