Skip to content
Snippets Groups Projects
Commit c681bd04 authored by Jean-Marc Valin's avatar Jean-Marc Valin
Browse files

Improved mode/channel/bandwidth control mechanism 

Now has tuning parameters for mono/stereo and voice/music. Also switches
to stereo during swb and without reducing the bandwidth.
parent bafbd08d
No related branches found
No related tags found
No related merge requests found
Hide whitespace changes
Inline Side-by-side
src/opus_encoder.c
+ 95
− 73
View file @ c681bd04
......@@ -83,23 +83,43 @@ struct OpusEncoder {
int rangeFinal;
};
/* Transition tables for the voice and audio modes. First column is the
/* Transition tables for the voice and music. First column is the
middle (memoriless) threshold. The second column is the hysteresis
(difference with the middle) */
static const int voice_bandwidth_thresholds[10] = {
static const opus_int32 mono_voice_bandwidth_thresholds[8] = {
11000, 1000, /* NB<->MB */
14000, 1000, /* MB<->WB */
21000, 2000, /* WB<->SWB */
29000, 2000, /* SWB<->FB */
};
static const int audio_bandwidth_thresholds[10] = {
30000, 0, /* MB not allowed */
20000, 2000, /* MB<->WB */
26000, 2000, /* WB<->SWB */
static const opus_int32 mono_music_bandwidth_thresholds[8] = {
14000, 1000, /* MB not allowed */
18000, 2000, /* MB<->WB */
24000, 2000, /* WB<->SWB */
33000, 2000, /* SWB<->FB */
};
static const opus_int32 stereo_voice_bandwidth_thresholds[8] = {
11000, 1000, /* NB<->MB */
14000, 1000, /* MB<->WB */
21000, 2000, /* WB<->SWB */
32000, 2000, /* SWB<->FB */
};
static const opus_int32 stereo_music_bandwidth_thresholds[8] = {
14000, 1000, /* MB not allowed */
18000, 2000, /* MB<->WB */
24000, 2000, /* WB<->SWB */
48000, 2000, /* SWB<->FB */
};
/* Threshold bit-rates for switching between mono and stereo */
static const opus_int32 stereo_voice_threshold = 26000;
static const opus_int32 stereo_music_threshold = 36000;
/* Threshold bit-rate for switching between SILK/hybrid and CELT-only */
static const opus_int32 mode_thresholds[2][2] = {
/* voice */ /* music */
{ 48000, 24000}, /* mono */
{ 48000, 24000}, /* stereo */
};
int opus_encoder_get_size(int channels)
{
int silkEncSizeBytes, celtEncSizeBytes;
......@@ -173,7 +193,7 @@ int opus_encoder_init(OpusEncoder* st, int Fs, int channels, int application)
st->application = application;
st->signal_type = OPUS_SIGNAL_AUTO;
st->user_bandwidth = OPUS_BANDWIDTH_AUTO;
st->voice_ratio = 90;
st->voice_ratio = -1;
st->encoder_buffer = st->Fs/100;
st->delay_compensation = st->Fs/400;
......@@ -350,9 +370,10 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
VARDECL(opus_val16, pcm_buf);
int nb_compr_bytes;
int to_celt = 0;
opus_int32 mono_rate;
opus_uint32 redundant_rng = 0;
int cutoff_Hz, hp_freq_smth1;
int voice_est;
opus_int32 equiv_rate;
ALLOC_STACK;
st->rangeFinal = 0;
......@@ -370,41 +391,45 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
else
st->bitrate_bps = st->user_bitrate_bps;
/* Equivalent 20-ms rate for mode/channel/bandwidth decisions */
equiv_rate = st->bitrate_bps - 60*(st->Fs/frame_size - 50);
if (st->signal_type == OPUS_SIGNAL_VOICE)
voice_est = 127;
else if (st->signal_type == OPUS_SIGNAL_MUSIC)
voice_est = 0;
else if (st->voice_ratio >= 0)
voice_est = st->voice_ratio*327>>8;
else if (st->application == OPUS_APPLICATION_VOIP)
voice_est = 115;
else
voice_est = 64;
#ifdef FUZZING
/* Random mono/stereo decision */
if (st->channels == 2 && (rand()&0x1F)==0)
st->stream_channels = 3-st->stream_channels;
#else
/* Rate-dependent mono-stereo decision */
if (st->force_mono)
{
st->stream_channels = 1;
} else if (st->mode == MODE_CELT_ONLY && st->channels == 2)
} else if (st->channels == 2)
{
opus_int32 decision_rate;
decision_rate = st->bitrate_bps + st->voice_ratio*st->voice_ratio;
/* Add some hysteresis */
if (st->stream_channels == 2)
decision_rate += 4000;
else
decision_rate -= 4000;
if (decision_rate>48000)
st->stream_channels = 2;
else
st->stream_channels = 1;
opus_int32 stereo_threshold;
stereo_threshold = stereo_music_threshold + ((voice_est*voice_est*(stereo_voice_threshold-stereo_music_threshold))>>14);
if (st->stream_channels == 2)
stereo_threshold -= 4000;
else
stereo_threshold += 4000;
st->stream_channels = (equiv_rate > stereo_threshold) ? 2 : 1;
} else {
st->stream_channels = st->channels;
}
#ifdef FUZZING
if (st->channels == 2 && (rand()&0x1F)==0)
st->stream_channels = 3-st->stream_channels;
#endif
/* Equivalent bit-rate for mono */
mono_rate = st->bitrate_bps;
if (st->stream_channels==2)
mono_rate = 2*mono_rate/3;
/* Compensate for smaller frame sizes assuming an equivalent overhead
of 60 bits/frame */
mono_rate -= 60*(st->Fs/frame_size - 50);
#ifdef FUZZING
/* Random mode switching */
if ((rand()&0xF)==0)
{
if ((rand()&0x1)==0)
......@@ -419,44 +444,26 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
}
#else
/* Mode selection depending on application and signal type */
if (st->application==OPUS_APPLICATION_VOIP)
{
opus_int32 threshold;
threshold = 20000;
/* OPUS_APPLICATION_VOIP default to auto high-pass */
/* Hysteresis */
if (st->prev_mode == MODE_CELT_ONLY)
threshold -= 4000;
else if (st->prev_mode>0)
threshold += 4000;
/* OPUS_APPLICATION_VOIP defaults to MODE_SILK_ONLY */
if (st->signal_type == OPUS_SIGNAL_MUSIC && mono_rate > threshold)
st->mode = MODE_CELT_ONLY;
else
st->mode = MODE_SILK_ONLY;
} else {/* OPUS_APPLICATION_AUDIO */
opus_int32 threshold;
/* SILK/CELT threshold is higher for voice than for music */
threshold = 36000;
/* OPUS_APPLICATION_AUDIO disables the high-pass */
if (st->signal_type == OPUS_SIGNAL_MUSIC)
threshold -= 20000;
else if (st->signal_type == OPUS_SIGNAL_VOICE)
threshold += 8000;
/* Hysteresis */
if (st->prev_mode == MODE_CELT_ONLY)
threshold -= 4000;
else if (st->prev_mode>0)
threshold += 4000;
if (mono_rate>threshold)
st->mode = MODE_CELT_ONLY;
else
st->mode = MODE_SILK_ONLY;
int chan;
opus_int32 mode_voice, mode_music;
opus_int32 threshold;
chan = (st->channels==2) && !st->force_mono;
mode_voice = mode_thresholds[chan][0];
mode_music = mode_thresholds[chan][1];
threshold = mode_music + ((voice_est*voice_est*(mode_voice-mode_music))>>14);
/* Hysteresis */
if (st->prev_mode == MODE_CELT_ONLY)
threshold -= 4000;
else if (st->prev_mode>0)
threshold += 4000;
st->mode = (equiv_rate >= threshold) ? MODE_CELT_ONLY: MODE_SILK_ONLY;
}
#endif
/* Override the chosen mode to make sure we meet the requested frame size */
if (st->mode == MODE_CELT_ONLY && frame_size > st->Fs/50)
st->mode = MODE_SILK_ONLY;
......@@ -491,10 +498,24 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
/* Automatic (rate-dependent) bandwidth selection */
if (st->mode == MODE_CELT_ONLY || st->first || st->silk_mode.allowBandwidthSwitch)
{
const int *bandwidth_thresholds;
const opus_int32 *voice_bandwidth_thresholds, *music_bandwidth_thresholds;
opus_int32 bandwidth_thresholds[8];
int bandwidth = OPUS_BANDWIDTH_FULLBAND;
bandwidth_thresholds = st->mode == MODE_CELT_ONLY ? audio_bandwidth_thresholds : voice_bandwidth_thresholds;
if (st->channels==2 && !st->force_mono)
{
voice_bandwidth_thresholds = stereo_voice_bandwidth_thresholds;
music_bandwidth_thresholds = stereo_music_bandwidth_thresholds;
} else {
voice_bandwidth_thresholds = mono_voice_bandwidth_thresholds;
music_bandwidth_thresholds = mono_music_bandwidth_thresholds;
}
/* Interpolate bandwidth thresholds depending on voice estimation */
for (i=0;i<8;i++)
{
bandwidth_thresholds[i] = music_bandwidth_thresholds[i]
+ ((voice_est*voice_est*(voice_bandwidth_thresholds[i]-music_bandwidth_thresholds[i]))>>14);
}
do {
int threshold, hysteresis;
threshold = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)];
......@@ -506,7 +527,7 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
else
threshold += hysteresis;
}
if (mono_rate >= threshold)
if (equiv_rate >= threshold)
break;
} while (--bandwidth>OPUS_BANDWIDTH_NARROWBAND);
st->bandwidth = bandwidth;
......@@ -545,6 +566,7 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
if (st->mode == MODE_HYBRID && st->bandwidth <= OPUS_BANDWIDTH_WIDEBAND)
st->mode = MODE_SILK_ONLY;
/* printf("%d %d %d %d\n", st->bitrate_bps, st->stream_channels, st->mode, st->bandwidth); */
bytes_target = st->bitrate_bps * frame_size / (st->Fs * 8) - 1;
data += 1;
......@@ -1040,7 +1062,7 @@ int opus_encoder_ctl(OpusEncoder *st, int request, ...)
case OPUS_SET_VOICE_RATIO_REQUEST:
{
opus_int32 value = va_arg(ap, opus_int32);
if (value>100 || value<0)
if (value>100 || value<-1)
goto bad_arg;
st->voice_ratio = value;
}
......
src/test_opus.c
+ 12
− 0
View file @ c681bd04
......@@ -116,6 +116,7 @@ int main(int argc, char *argv[])
int encode_only=0, decode_only=0;
int max_frame_size = 960*6;
int curr_read=0;
int sweep_bps = 0;
if (argc < 7 )
{
......@@ -223,6 +224,9 @@ int main(int argc, char *argv[])
} else if( STR_CASEINSENSITIVE_COMPARE( argv[ args ], "-loss" ) == 0 ) {
packet_loss_perc = atoi( argv[ args + 1 ] );
args += 2;
} else if( STR_CASEINSENSITIVE_COMPARE( argv[ args ], "-sweep" ) == 0 ) {
sweep_bps = atoi( argv[ args + 1 ] );
args += 2;
} else {
printf( "Error: unrecognized setting: %s\n\n", argv[ args ] );
print_usage( argv );
......@@ -363,6 +367,14 @@ int main(int argc, char *argv[])
}
len[toggle] = opus_encode(enc, in, frame_size, data[toggle], max_payload_bytes);
if (sweep_bps!=0)
{
bitrate_bps += sweep_bps;
/* safety */
if (bitrate_bps<1000)
bitrate_bps = 1000;
opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate_bps));
}
opus_encoder_ctl(enc, OPUS_GET_FINAL_RANGE(&enc_final_range[toggle]));
if (len[toggle] < 0)
{
......
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment