Making use of the opus_int* types in the toplevel Opus code
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4 changed files with 20 additions and 18 deletions
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@ -28,6 +28,8 @@
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#ifndef OPUS_H
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#define OPUS_H
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#include "opus_types.h"
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#ifdef __cplusplus
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extern "C" {
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#endif
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@ -175,7 +177,7 @@ OPUS_EXPORT OpusEncoder *opus_encoder_init(
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/* Returns length of the data payload (in bytes) */
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OPUS_EXPORT int opus_encode(
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OpusEncoder *st, /* Encoder state */
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const short *pcm, /* Input signal (interleaved if 2 channels). length is frame_size*channels */
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const opus_int16 *pcm, /* Input signal (interleaved if 2 channels). length is frame_size*channels */
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int frame_size, /* Number of samples per frame of input signal */
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unsigned char *data, /* Output payload (no more than max_data_bytes long) */
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int max_data_bytes /* Allocated memory for payload; don't use for controlling bitrate */
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@ -200,7 +202,7 @@ OPUS_EXPORT int opus_decode(
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OpusDecoder *st, /* Decoder state */
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const unsigned char *data, /* Input payload. Use a NULL pointer to indicate packet loss */
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int len, /* Number of bytes in payload */
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short *pcm, /* Output signal (interleaved if 2 channels). length is frame_size*channels */
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opus_int16 *pcm, /* Output signal (interleaved if 2 channels). length is frame_size*channels */
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int frame_size, /* Number of samples per frame of input signal */
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int decode_fec /* Flag (0/1) to request that any in-band forward error correction data be */
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/* decoded. If no such data is available the frame is decoded as if it were lost. */
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@ -111,7 +111,7 @@ OpusDecoder *opus_decoder_create(int Fs, int channels)
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return opus_decoder_init((OpusDecoder*)raw_state, Fs, channels);
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}
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static void smooth_fade(const short *in1, const short *in2, short *out,
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static void smooth_fade(const opus_int16 *in1, const opus_int16 *in2, opus_int16 *out,
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int overlap, int channels, const opus_val16 *window, int Fs)
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{
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int i, c;
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@ -144,7 +144,7 @@ static int opus_packet_get_mode(const unsigned char *data)
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}
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static int opus_decode_frame(OpusDecoder *st, const unsigned char *data,
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int len, short *pcm, int frame_size, int decode_fec)
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int len, opus_int16 *pcm, int frame_size, int decode_fec)
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{
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void *silk_dec;
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CELTDecoder *celt_dec;
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@ -152,8 +152,8 @@ static int opus_decode_frame(OpusDecoder *st, const unsigned char *data,
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ec_dec dec;
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silk_DecControlStruct DecControl;
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opus_int32 silk_frame_size;
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short pcm_celt[960*2];
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short pcm_transition[480*2];
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opus_int16 pcm_celt[960*2];
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opus_int16 pcm_transition[480*2];
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int audiosize;
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int mode;
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@ -162,7 +162,7 @@ static int opus_decode_frame(OpusDecoder *st, const unsigned char *data,
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int redundancy=0;
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int redundancy_bytes = 0;
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int celt_to_silk=0;
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short redundant_audio[240*2];
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opus_int16 redundant_audio[240*2];
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int c;
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int F2_5, F5, F10, F20;
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const opus_val16 *window;
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@ -413,7 +413,7 @@ static int parse_size(const unsigned char *data, int len, short *size)
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}
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int opus_decode(OpusDecoder *st, const unsigned char *data,
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int len, short *pcm, int frame_size, int decode_fec)
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int len, opus_int16 *pcm, int frame_size, int decode_fec)
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{
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int i, bytes, nb_samples;
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int count;
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@ -48,8 +48,8 @@ struct OpusDecoder {
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int rangeFinal;
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};
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static inline short SAT16(int x) {
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return x > 32767 ? 32767 : x < -32768 ? -32768 : (short)x;
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static inline opus_int16 SAT16(opus_int32 x) {
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return x > 32767 ? 32767 : x < -32768 ? -32768 : (opus_int16)x;
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};
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#endif /* OPUS_DECODER_H */
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@ -162,7 +162,7 @@ OpusEncoder *opus_encoder_create(int Fs, int channels, int mode)
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return opus_encoder_init((OpusEncoder*)raw_state, Fs, channels, mode);
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}
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int opus_encode(OpusEncoder *st, const short *pcm, int frame_size,
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int opus_encode(OpusEncoder *st, const opus_int16 *pcm, int frame_size,
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unsigned char *data, int max_data_bytes)
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{
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void *silk_enc;
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@ -181,7 +181,7 @@ int opus_encode(OpusEncoder *st, const short *pcm, int frame_size,
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int celt_to_silk = 0;
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/* TODO: This is 60 only so we can handle 60ms speech/audio switching
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it shouldn't be too hard to reduce to 20 ms if needed */
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short pcm_buf[60*48*2];
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opus_int16 pcm_buf[60*48*2];
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int nb_compr_bytes;
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int to_celt = 0;
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opus_int32 mono_rate;
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@ -510,16 +510,16 @@ int opus_encode(OpusEncoder *st, const short *pcm, int frame_size,
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delta_Q14 = ( st->hybrid_stereo_width_Q14 - st->silk_mode.stereoWidth_Q14 ) / nSamples_8ms;
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for( i = 0; i < nSamples_8ms; i++ ) {
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width_Q14 += delta_Q14;
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diff = pcm_buf[ 2*i+1 ] - (int)pcm_buf[ 2*i ];
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diff = pcm_buf[ 2*i+1 ] - (opus_int32)pcm_buf[ 2*i ];
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diff = ( diff * width_Q14 ) >> 15;
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pcm_buf[ 2*i ] = (short)( pcm_buf[ 2*i ] + diff );
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pcm_buf[ 2*i+1 ] = (short)( pcm_buf[ 2*i+1 ] - diff );
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pcm_buf[ 2*i ] = (opus_int16)( pcm_buf[ 2*i ] + diff );
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pcm_buf[ 2*i+1 ] = (opus_int16)( pcm_buf[ 2*i+1 ] - diff );
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}
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for( ; i < frame_size; i++ ) {
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diff = pcm_buf[ 2*i+1 ] - (int)pcm_buf[ 2*i ];
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diff = pcm_buf[ 2*i+1 ] - (opus_int32)pcm_buf[ 2*i ];
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diff = ( diff * width_Q14 ) >> 15;
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pcm_buf[ 2*i ] = (short)( pcm_buf[ 2*i ] + diff );
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pcm_buf[ 2*i+1 ] = (short)( pcm_buf[ 2*i+1 ] - diff );
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pcm_buf[ 2*i ] = (opus_int16)( pcm_buf[ 2*i ] + diff );
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pcm_buf[ 2*i+1 ] = (opus_int16)( pcm_buf[ 2*i+1 ] - diff );
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}
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st->hybrid_stereo_width_Q14 = st->silk_mode.stereoWidth_Q14;
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}
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