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Fixes a problem with the SILK toMono when we're encoding 40/60 ms frames

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This commit is contained in:
Jean-Marc Valin 2011-10-07 22:29:38 -04:00
parent 6e5f6d048a
commit 1e22a4b7e9
1 changed files with 28 additions and 18 deletions
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20
src/opus_encoder.c
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@ -462,16 +462,17 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
else else
voice_est = 64; voice_est = 64;
if (st->force_channels!=OPUS_AUTO && st->channels == 2)
{
st->stream_channels = st->force_channels;
} else {
#ifdef FUZZING #ifdef FUZZING
/* Random mono/stereo decision */ /* Random mono/stereo decision */
if (st->channels == 2 && (rand()&0x1F)==0) if (st->channels == 2 && (rand()&0x1F)==0)
st->stream_channels = 3-st->stream_channels; st->stream_channels = 3-st->stream_channels;
#else #else
/* Rate-dependent mono-stereo decision */ /* Rate-dependent mono-stereo decision */
if (st->force_channels!=OPUS_AUTO && st->channels == 2) if (st->channels == 2)
{
st->stream_channels = st->force_channels;
} else if (st->channels == 2)
{ {
opus_int32 stereo_threshold; opus_int32 stereo_threshold;
stereo_threshold = stereo_music_threshold + ((voice_est*voice_est*(stereo_voice_threshold-stereo_music_threshold))>>14); stereo_threshold = stereo_music_threshold + ((voice_est*voice_est*(stereo_voice_threshold-stereo_music_threshold))>>14);
@ -484,6 +485,7 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
st->stream_channels = st->channels; st->stream_channels = st->channels;
} }
#endif #endif
}
if (st->stream_channels == 1 && st->prev_channels ==2 && st->silk_mode.toMono==0) if (st->stream_channels == 1 && st->prev_channels ==2 && st->silk_mode.toMono==0)
{ {
@ -628,10 +630,11 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
VARDECL(unsigned char, tmp_data); VARDECL(unsigned char, tmp_data);
VARDECL(unsigned char, rp_); VARDECL(unsigned char, rp_);
int nb_frames; int nb_frames;
int bak_mode, bak_bandwidth, bak_channels; int bak_mode, bak_bandwidth, bak_channels, bak_to_mono;
OpusRepacketizer *rp; OpusRepacketizer *rp;
int bytes_per_frame; int bytes_per_frame;
nb_frames = frame_size > st->Fs/25 ? 3 : 2; nb_frames = frame_size > st->Fs/25 ? 3 : 2;
bytes_per_frame = max_data_bytes/nb_frames-3; bytes_per_frame = max_data_bytes/nb_frames-3;
@ -647,10 +650,16 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
st->user_forced_mode = st->mode; st->user_forced_mode = st->mode;
st->user_bandwidth = st->bandwidth; st->user_bandwidth = st->bandwidth;
st->force_channels = st->stream_channels; st->force_channels = st->stream_channels;
bak_to_mono = st->silk_mode.toMono;
if (bak_to_mono)
st->force_channels = 1;
else
st->prev_channels = st->stream_channels;
for (i=0;i<nb_frames;i++) for (i=0;i<nb_frames;i++)
{ {
int tmp_len; int tmp_len;
st->silk_mode.toMono = 0;
tmp_len = opus_encode_native(st, pcm+i*(st->channels*st->Fs/50), st->Fs/50, tmp_data+i*bytes_per_frame, bytes_per_frame); tmp_len = opus_encode_native(st, pcm+i*(st->channels*st->Fs/50), st->Fs/50, tmp_data+i*bytes_per_frame, bytes_per_frame);
ret = opus_repacketizer_cat(rp, tmp_data+i*bytes_per_frame, tmp_len); ret = opus_repacketizer_cat(rp, tmp_data+i*bytes_per_frame, tmp_len);
} }
@ -659,6 +668,7 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
st->user_forced_mode = bak_mode; st->user_forced_mode = bak_mode;
st->user_bandwidth = bak_bandwidth; st->user_bandwidth = bak_bandwidth;
st->force_channels = bak_channels; st->force_channels = bak_channels;
st->silk_mode.toMono = bak_to_mono;
RESTORE_STACK; RESTORE_STACK;
return ret; return ret;
} }