diff options
Diffstat (limited to 'src/liblame_wrapper.cc')
| -rw-r--r-- | src/liblame_wrapper.cc | 127 |
1 files changed, 51 insertions, 76 deletions
diff --git a/src/liblame_wrapper.cc b/src/liblame_wrapper.cc index 7af9b68..90b8824 100644 --- a/src/liblame_wrapper.cc +++ b/src/liblame_wrapper.cc @@ -31,6 +31,9 @@ /* * $Log$ + * Revision 1.3 2005/07/07 12:42:19 deva + * *** empty log message *** + * * Revision 1.2 2005/07/05 23:15:16 deva * *** empty log message *** * @@ -62,6 +65,18 @@ LibLAMEWrapper::LibLAMEWrapper(Info *i) lame_set_mode(gfp, STEREO); lame_set_brate(gfp, 192); lame_set_strict_ISO(gfp, 1); + lame_set_num_samples(gfp, SAMPLES); + + lame_set_copyright(gfp, 0); // is there a copyright on the encoded data? + lame_set_original(gfp, 1); // is the encoded data on the original media? + lame_set_error_protection(gfp, 0);// add 2 byte CRC protection to each frame? + lame_set_padding_type(gfp, PAD_NO); // PAD_NO, PAD_ALL, PAD_ADJUST, PAD_MAX_INDICATOR + // 0 = do not pad frames + // 1 = always pad frames + // 2 = adjust padding to satisfy bit rate + lame_set_extension(gfp, 0); // private extension bit + + info->info("Lame version %d", lame_get_version(gfp)); if (lame_init_params(gfp) < 0) { info->error("LAME parameter initialization failed."); @@ -72,8 +87,6 @@ LibLAMEWrapper::LibLAMEWrapper(Info *i) audio_buffer[0] = new int16_t[AUDIO_BUFFER_SIZE]; audio_buffer[1] = new int16_t[AUDIO_BUFFER_SIZE]; - audio_buffer[2] = new int16_t[AUDIO_BUFFER_SIZE]; - audio_buffer[3] = new int16_t[AUDIO_BUFFER_SIZE]; // And now for the dv decoder! decoder = NULL; @@ -85,8 +98,6 @@ LibLAMEWrapper::~LibLAMEWrapper() delete audio_buffer[0]; delete audio_buffer[1]; - delete audio_buffer[2]; - delete audio_buffer[3]; } Frame *LibLAMEWrapper::encode(Frame *dvframe) @@ -104,66 +115,53 @@ Frame *LibLAMEWrapper::encode(Frame *dvframe) decoder->num_dif_seqs = 12; } - // See - // http://www.koders.com/cpp/fidE614E999154E2B4A813DA272C4421633063C78CA.aspx - // line 769 - /** * Decode audio using libdv */ - // int n, i; - // int16_t* s = ( int16_t * ) sound; - dv_decode_full_audio( decoder, dvframe->data, audio_buffer ); - /* - memset(audio_buffer[0], 0, sizeof(audio_buffer[0])); - memset(audio_buffer[1], 0, sizeof(audio_buffer[1])); - */ - /* - for ( n = 0; n < SAMPLES; ++n ) - for ( i = 0; i < CHANNELS; i++ ) - *s++ = audio_buffer[ i ][ n ]; - */ - -/* - * input pcm data, output (maybe) mp3 frames. - * This routine handles all buffering, resampling and filtering for you. - * - * The required mp3buf_size can be computed from num_samples, - * samplerate and encoding rate, but here is a worst case estimate: - * - * return code number of bytes output in mp3buffer. can be 0 - * if return code = -1: mp3buffer was too small - * - * mp3buf_size in bytes = 1.25*num_samples + 7200 - * - * I think a tighter bound could be: (mt, March 2000) - * MPEG1: - * num_samples*(bitrate/8)/samplerate + 4*1152*(bitrate/8)/samplerate + 512 - * MPEG2: - * num_samples*(bitrate/8)/samplerate + 4*576*(bitrate/8)/samplerate + 256 - * - * but test first if you use that! - * - * set mp3buf_size = 0 and LAME will not check if mp3buf_size is - * large enough. - * - * NOTE: - * if gfp->num_channels=2, but gfp->mode = 3 (mono), the L & R channels - * will be averaged into the L channel before encoding only the L channel - * This will overwrite the data in buffer_l[] and buffer_r[]. - * -*/ + /** + * input pcm data, output (maybe) mp3 frames. + * This routine handles all buffering, resampling and filtering for you. + * + * The required mp3buf_size can be computed from num_samples, + * samplerate and encoding rate, but here is a worst case estimate: + * + * return code number of bytes output in mp3buffer. can be 0 + * if return code = -1: mp3buffer was too small + * + * mp3buf_size in bytes = 1.25*num_samples + 7200 + * + * I think a tighter bound could be: (mt, March 2000) + * MPEG1: + * num_samples*(bitrate/8)/samplerate + 4*1152*(bitrate/8)/samplerate + 512 + * MPEG2: + * num_samples*(bitrate/8)/samplerate + 4*576*(bitrate/8)/samplerate + 256 + * + * but test first if you use that! + * + * set mp3buf_size = 0 and LAME will not check if mp3buf_size is + * large enough. + * + * NOTE: + * if gfp->num_channels=2, but gfp->mode = 3 (mono), the L & R channels + * will be averaged into the L channel before encoding only the L channel + * This will overwrite the data in buffer_l[] and buffer_r[]. + * + */ Frame* audio_frame = new Frame(NULL, (int)(1.25 * SAMPLES + 7200)); - const short int *buffer_l = audio_buffer[0]; // PCM data for left channel - const short int *buffer_r = audio_buffer[1]; // PCM data for right channel + const short int *buffer_l = audio_buffer[0]; // PCM data for left channel + const short int *buffer_r = audio_buffer[1]; // PCM data for right channel const int nsamples = SAMPLES; // number of samples per channel unsigned char* mp3buf = audio_frame->data; // pointer to encoded MP3 stream const int mp3buf_size = audio_frame->size; // number of valid octets in this - int val = lame_encode_buffer(gfp, buffer_l, buffer_r, nsamples, mp3buf, mp3buf_size); + int val; + val = lame_encode_buffer(gfp, buffer_l, buffer_r, nsamples, mp3buf, mp3buf_size); + // val = lame_encode_mp3_frame(gfp, buffer_l, buffer_r, mp3buf, mp3buf_size); + + info->info("Framenr: %d", lame_get_frameNum(gfp)); if(val < 0) { switch(val) { @@ -185,29 +183,6 @@ Frame *LibLAMEWrapper::encode(Frame *dvframe) } } - /* - val = lame_encode_flush(gfp, mp3buf, val); - - if(val < 0) { - switch(val) { - case -1: // mp3buf was too small - info->error("Lame encoding failed (during flush), mp3buf was too small."); - break; - case -2: // malloc() problem - info->error("Lame encoding failed (during flush), due to malloc() problem."); - break; - case -3: // lame_init_params() not called - info->error("Lame encoding failed (during flush), lame_init_params() not called."); - break; - case -4: // psycho acoustic problems - info->error("Lame encoding failed (during flush), due to psycho acoustic problems."); - break; - default: - info->error("Lame encoding failed (during flush), due to unknown error."); - break; - } - } - */ audio_frame->size = val; return audio_frame; |