diff options
Diffstat (limited to 'libhb')
-rw-r--r-- | libhb/audio_remap.c | 22 | ||||
-rw-r--r-- | libhb/audio_remap.h | 3 | ||||
-rw-r--r-- | libhb/audio_resample.c | 4 | ||||
-rw-r--r-- | libhb/audio_resample.h | 5 | ||||
-rw-r--r-- | libhb/deca52.c | 11 | ||||
-rw-r--r-- | libhb/decavcodec.c | 43 | ||||
-rw-r--r-- | libhb/declpcm.c | 3 | ||||
-rw-r--r-- | libhb/hb.c | 17 | ||||
-rw-r--r-- | libhb/hbffmpeg.h | 1 | ||||
-rw-r--r-- | libhb/platform/macosx/encca_aac.c | 2 |
10 files changed, 31 insertions, 80 deletions
diff --git a/libhb/audio_remap.c b/libhb/audio_remap.c index 44f0d73d3..d8c7f3de7 100644 --- a/libhb/audio_remap.c +++ b/libhb/audio_remap.c @@ -246,19 +246,25 @@ fail: } void hb_audio_remap_set_channel_layout(hb_audio_remap_t *remap, - uint64_t channel_layout, - int channels) + uint64_t channel_layout) { if (remap != NULL) { int ii; remap->remap_needed = 0; + // sanitize the layout + if (channel_layout == AV_CH_LAYOUT_STEREO_DOWNMIX) + { + channel_layout = AV_CH_LAYOUT_STEREO; + } + remap->nchannels = av_get_channel_layout_nb_channels(channel_layout); + // in some cases, remapping is not necessary and/or supported - if (channels > HB_AUDIO_REMAP_MAX_CHANNELS) + if (remap->nchannels > HB_AUDIO_REMAP_MAX_CHANNELS) { hb_log("hb_audio_remap_set_channel_layout: too many channels (%d)", - channels); + remap->nchannels); return; } if (remap->channel_map_in == remap->channel_map_out) @@ -266,14 +272,6 @@ void hb_audio_remap_set_channel_layout(hb_audio_remap_t *remap, return; } - // sanitize the layout - channel_layout = hb_ff_layout_xlat(channel_layout, channels); - if (channel_layout == AV_CH_LAYOUT_STEREO_DOWNMIX) - { - channel_layout = AV_CH_LAYOUT_STEREO; - } - remap->nchannels = av_get_channel_layout_nb_channels(channel_layout); - // build the table and check whether remapping is necessary hb_audio_remap_build_table(remap->channel_map_out, remap->channel_map_in, channel_layout, diff --git a/libhb/audio_remap.h b/libhb/audio_remap.h index 3092fb854..6f44ad6b9 100644 --- a/libhb/audio_remap.h +++ b/libhb/audio_remap.h @@ -73,8 +73,7 @@ hb_audio_remap_t* hb_audio_remap_init(enum AVSampleFormat sample_fmt, * Must be called at least once before remapping. */ void hb_audio_remap_set_channel_layout(hb_audio_remap_t *remap, - uint64_t channel_layout, - int channels); + uint64_t channel_layout); /* * Free an hb_audio_remap_t. diff --git a/libhb/audio_resample.c b/libhb/audio_resample.c index b48cc7271..913186cc0 100644 --- a/libhb/audio_resample.c +++ b/libhb/audio_resample.c @@ -75,12 +75,10 @@ fail: } void hb_audio_resample_set_channel_layout(hb_audio_resample_t *resample, - uint64_t channel_layout, - int channels) + uint64_t channel_layout) { if (resample != NULL) { - channel_layout = hb_ff_layout_xlat(channel_layout, channels); if (channel_layout == AV_CH_LAYOUT_STEREO_DOWNMIX) { // Dolby Surround is Stereo when it comes to remixing diff --git a/libhb/audio_resample.h b/libhb/audio_resample.h index bac3b3292..9242d78a5 100644 --- a/libhb/audio_resample.h +++ b/libhb/audio_resample.h @@ -74,13 +74,10 @@ hb_audio_resample_t* hb_audio_resample_init(enum AVSampleFormat sample_fmt, * * They should be called whenever the relevant characteristic(s) differ from the * requested output characteristics, or if they may have changed in the source. - * - * Note: channel_layout is automatically sanitized. */ void hb_audio_resample_set_channel_layout(hb_audio_resample_t *resample, - uint64_t channel_layout, - int channels); + uint64_t channel_layout); void hb_audio_resample_set_mix_levels(hb_audio_resample_t *resample, double surround_mix_level, diff --git a/libhb/deca52.c b/libhb/deca52.c index f5dac1f96..5dde0378a 100644 --- a/libhb/deca52.c +++ b/libhb/deca52.c @@ -38,7 +38,6 @@ struct hb_work_private_s uint8_t buf[6][6][256 * sizeof(float)]; // decoded frame (up to 6 channels, 6 blocks * 256 samples) uint8_t *samples[6]; // pointers to the start of each plane (1 per channel) - int nchannels; int use_mix_levels; uint64_t channel_layout; hb_audio_remap_t *remap; @@ -383,13 +382,10 @@ static hb_buffer_t* Decode(hb_work_object_t *w) if (new_layout != pv->channel_layout) { pv->channel_layout = new_layout; - pv->nchannels = av_get_channel_layout_nb_channels(new_layout); hb_audio_remap_set_channel_layout(pv->remap, - pv->channel_layout, - pv->nchannels); + pv->channel_layout); hb_audio_resample_set_channel_layout(pv->resample, - pv->channel_layout, - pv->nchannels); + pv->channel_layout); } if (pv->use_mix_levels) { @@ -416,7 +412,8 @@ static hb_buffer_t* Decode(hb_work_object_t *w) * * copy samples to our internal buffer */ - for (j = 0; j < pv->nchannels; j++) + for (j = 0; + j < av_get_channel_layout_nb_channels(pv->channel_layout); j++) { pv->samples[j] = (uint8_t*)pv->buf[j]; memcpy(pv->buf[j][i], block_samples, 256 * sizeof(float)); diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c index a0f5a59c4..5a4b0513c 100644 --- a/libhb/decavcodec.c +++ b/libhb/decavcodec.c @@ -655,18 +655,17 @@ static int decavcodecaBSInfo( hb_work_object_t *w, const hb_buffer_t *buf, avp.size = pbuffer_size; len = avcodec_decode_audio4(context, frame, &got_frame, &avp); - if (len > 0 && context->sample_rate > 0) + if (len > 0 && got_frame) { info->rate_base = 1; - info->rate = context->sample_rate; + info->rate = frame->sample_rate; info->samples_per_frame = frame->nb_samples; int bps = av_get_bits_per_sample(context->codec_id); - if (bps > 0 && context->channels > 0) + int channels = av_get_channel_layout_nb_channels(frame->channel_layout); + if (bps > 0) { - info->bitrate = (bps * - context->channels * - context->sample_rate); + info->bitrate = (bps * channels * info->rate); } else if (context->bit_rate > 0) { @@ -683,9 +682,7 @@ static int decavcodecaBSInfo( hb_work_object_t *w, const hb_buffer_t *buf, } else { - info->channel_layout = - hb_ff_layout_xlat(context->channel_layout, - context->channels); + info->channel_layout = frame->channel_layout; } ret = 1; @@ -2081,26 +2078,11 @@ static void decodeAudio(hb_audio_t *audio, hb_work_private_t *pv, uint8_t *data, pos += len; - // set the duration on every frame since the stream format can - // change (it shouldn't but there's no way to guarantee it). - // duration is a scaling factor to go from #bytes in the decoded - // frame to frame time (in 90KHz mpeg ticks). 'channels' converts - // total samples to per-channel samples. 'sample_rate' converts - // per-channel samples to seconds per sample and the 90000 - // is mpeg ticks per second. - // - // Also, sample rate is not available until we have decoded some - // audio. - if ( pv->context->sample_rate ) - { - pv->duration = 90000. / (double)( pv->context->sample_rate ); - } - if (got_frame) { hb_buffer_t *out; - double duration = pv->frame->nb_samples * pv->duration; - double pts_next = pv->pts_next + duration; + double duration = (90000. * pv->frame->nb_samples / + (double)pv->frame->sample_rate); if (audio->config.out.codec & HB_ACODEC_PASS_FLAG) { @@ -2113,10 +2095,9 @@ static void decodeAudio(hb_audio_t *audio, hb_work_private_t *pv, uint8_t *data, else { hb_audio_resample_set_channel_layout(pv->resample, - context->channel_layout, - context->channels); + pv->frame->channel_layout); hb_audio_resample_set_sample_fmt(pv->resample, - context->sample_fmt); + pv->frame->format); if (hb_audio_resample_update(pv->resample)) { hb_log("decavcodec: hb_audio_resample_update() failed"); @@ -2132,8 +2113,8 @@ static void decodeAudio(hb_audio_t *audio, hb_work_private_t *pv, uint8_t *data, { out->s.start = pv->pts_next; out->s.duration = duration; - out->s.stop = pts_next; - pv->pts_next = pts_next; + out->s.stop = duration + pv->pts_next; + pv->pts_next = duration + pv->pts_next; hb_list_add(pv->list, out); } } diff --git a/libhb/declpcm.c b/libhb/declpcm.c index 7a17f2a14..a8f2e32d4 100644 --- a/libhb/declpcm.c +++ b/libhb/declpcm.c @@ -330,8 +330,7 @@ static hb_buffer_t *Decode( hb_work_object_t *w ) } hb_audio_resample_set_channel_layout(pv->resample, - hdr2layout[pv->nchannels - 1], - pv->nchannels); + hdr2layout[pv->nchannels - 1]); if (hb_audio_resample_update(pv->resample)) { hb_log("declpcm: hb_audio_resample_update() failed"); diff --git a/libhb/hb.c b/libhb/hb.c index 2b9609aa6..156d1241d 100644 --- a/libhb/hb.c +++ b/libhb/hb.c @@ -288,23 +288,6 @@ uint64_t hb_ff_mixdown_xlat(int hb_mixdown, int *downmix_mode) return ff_layout; } -uint64_t hb_ff_layout_xlat(uint64_t ff_channel_layout, int nchannels) -{ - uint64_t hb_layout = ff_channel_layout; - if (!hb_layout || av_get_channel_layout_nb_channels(hb_layout) != nchannels) - { - hb_layout = av_get_default_channel_layout(nchannels); - if (!hb_layout) - { - // This will likely not sound very good ;) - hb_layout = AV_CH_LAYOUT_STEREO; - hb_error("hb_ff_layout_xlat: unsupported layout 0x%"PRIx64" with %d channels", - ff_channel_layout, nchannels); - } - } - return hb_layout; -} - /* * Set sample format to the request format if supported by the codec. * The planar/packed variant of the requested format is the next best thing. diff --git a/libhb/hbffmpeg.h b/libhb/hbffmpeg.h index 776eec61a..94b1a903e 100644 --- a/libhb/hbffmpeg.h +++ b/libhb/hbffmpeg.h @@ -23,7 +23,6 @@ void hb_avcodec_init(void); int hb_avcodec_open(AVCodecContext *, AVCodec *, AVDictionary **, int); int hb_avcodec_close(AVCodecContext *); -uint64_t hb_ff_layout_xlat(uint64_t ff_channel_layout, int nchannels); uint64_t hb_ff_mixdown_xlat(int hb_mixdown, int *downmix_mode); void hb_ff_set_sample_fmt(AVCodecContext *, AVCodec *, enum AVSampleFormat); diff --git a/libhb/platform/macosx/encca_aac.c b/libhb/platform/macosx/encca_aac.c index 19e182a22..882b97a5a 100644 --- a/libhb/platform/macosx/encca_aac.c +++ b/libhb/platform/macosx/encca_aac.c @@ -297,7 +297,7 @@ int encCoreAudioInit(hb_work_object_t *w, hb_job_t *job, enum AAC_MODE mode) hb_error("encCoreAudioInit: hb_audio_remap_init() failed"); } uint64_t layout = hb_ff_mixdown_xlat(audio->config.out.mixdown, NULL); - hb_audio_remap_set_channel_layout(pv->remap, layout, pv->nchannels); + hb_audio_remap_set_channel_layout(pv->remap, layout); // get maximum output size AudioConverterGetProperty(pv->converter, |