diff options
Diffstat (limited to 'libhb/sync.c')
| -rw-r--r-- | libhb/sync.c | 116 |
1 files changed, 75 insertions, 41 deletions
diff --git a/libhb/sync.c b/libhb/sync.c index 7061aa1da..19ee605f0 100644 --- a/libhb/sync.c +++ b/libhb/sync.c @@ -47,6 +47,8 @@ typedef struct /* AC-3 */ int ac3_size; uint8_t * ac3_buf; + + double gain_factor; } hb_sync_audio_t; typedef struct @@ -1024,6 +1026,8 @@ hb_work_object_t hb_sync_audio = syncAudioClose }; +#define LVL_PLUS1DB 1.122462048 + static void InitAudio( hb_job_t * job, hb_sync_common_t * common, int i ) { hb_work_object_t * w; @@ -1101,6 +1105,9 @@ static void InitAudio( hb_job_t * job, hb_sync_common_t * common, int i ) w->audio->config.out.mixdown), &error ); sync->data.end_of_input = 0; } + + sync->gain_factor = pow(LVL_PLUS1DB, w->audio->config.out.gain); + hb_list_add( job->list_work, w ); } @@ -1110,54 +1117,81 @@ static hb_buffer_t * OutputAudioFrame( hb_audio_t *audio, hb_buffer_t *buf, int64_t start = (int64_t)sync->next_start; double duration = buf->stop - buf->start; - if( audio->config.in.samplerate == audio->config.out.samplerate || - audio->config.out.codec == HB_ACODEC_AC3_PASS || - audio->config.out.codec == HB_ACODEC_DCA_PASS ) - { - /* - * If we don't have to do sample rate conversion or this audio is - * pass-thru just send the input buffer downstream after adjusting - * its timestamps to make the output stream continuous. - */ - } - else + if ( !( audio->config.out.codec & HB_ACODEC_PASS_FLAG ) ) { - /* Not pass-thru - do sample rate conversion */ - int count_in, count_out; - hb_buffer_t * buf_raw = buf; - int channel_count = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown) * - sizeof( float ); + // Audio is not passthru. Check if we need to modify the audio + // in any way. + if( audio->config.in.samplerate != audio->config.out.samplerate ) + { + /* do sample rate conversion */ + int count_in, count_out; + hb_buffer_t * buf_raw = buf; + int channel_count = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown) * + sizeof( float ); - count_in = buf_raw->size / channel_count; - /* - * When using stupid rates like 44.1 there will always be some - * truncation error. E.g., a 1536 sample AC3 frame will turn into a - * 1536*44.1/48.0 = 1411.2 sample frame. If we just truncate the .2 - * the error will build up over time and eventually the audio will - * substantially lag the video. libsamplerate will keep track of the - * fractional sample & give it to us when appropriate if we give it - * an extra sample of space in the output buffer. - */ - count_out = ( duration * audio->config.out.samplerate ) / 90000 + 1; + count_in = buf_raw->size / channel_count; + /* + * When using stupid rates like 44.1 there will always be some + * truncation error. E.g., a 1536 sample AC3 frame will turn into a + * 1536*44.1/48.0 = 1411.2 sample frame. If we just truncate the .2 + * the error will build up over time and eventually the audio will + * substantially lag the video. libsamplerate will keep track of the + * fractional sample & give it to us when appropriate if we give it + * an extra sample of space in the output buffer. + */ + count_out = ( duration * audio->config.out.samplerate ) / 90000 + 1; - sync->data.input_frames = count_in; - sync->data.output_frames = count_out; - sync->data.src_ratio = (double)audio->config.out.samplerate / - (double)audio->config.in.samplerate; + sync->data.input_frames = count_in; + sync->data.output_frames = count_out; + sync->data.src_ratio = (double)audio->config.out.samplerate / + (double)audio->config.in.samplerate; - buf = hb_buffer_init( count_out * channel_count ); - sync->data.data_in = (float *) buf_raw->data; - sync->data.data_out = (float *) buf->data; - if( src_process( sync->state, &sync->data ) ) + buf = hb_buffer_init( count_out * channel_count ); + sync->data.data_in = (float *) buf_raw->data; + sync->data.data_out = (float *) buf->data; + if( src_process( sync->state, &sync->data ) ) + { + /* XXX If this happens, we're screwed */ + hb_log( "sync: audio 0x%x src_process failed", audio->id ); + } + hb_buffer_close( &buf_raw ); + + buf->size = sync->data.output_frames_gen * channel_count; + duration = (double)( sync->data.output_frames_gen * 90000 ) / + audio->config.out.samplerate; + } + if( audio->config.out.gain > 0.0 ) { - /* XXX If this happens, we're screwed */ - hb_log( "sync: audio 0x%x src_process failed", audio->id ); + int count, ii; + + count = buf->size / sizeof(float); + for ( ii = 0; ii < count; ii++ ) + { + double sample; + + sample = (double)*(((float*)buf->data)+ii); + sample *= sync->gain_factor; + if (sample > 0) + sample = MIN(sample, 32767.0); + else + sample = MAX(sample, -32768.0); + *(((float*)buf->data)+ii) = sample; + } } - hb_buffer_close( &buf_raw ); + else if( audio->config.out.gain < 0.0 ) + { + int count, ii; - buf->size = sync->data.output_frames_gen * channel_count; - duration = (double)( sync->data.output_frames_gen * 90000 ) / - audio->config.out.samplerate; + count = buf->size / sizeof(float); + for ( ii = 0; ii < count; ii++ ) + { + double sample; + + sample = (double)*(((float*)buf->data)+ii); + sample *= sync->gain_factor; + *(((float*)buf->data)+ii) = sample; + } + } } buf->frametype = HB_FRAME_AUDIO; buf->start = start; |