diff options
Diffstat (limited to 'audio')
-rw-r--r-- | audio/decode/ad_lavc.c | 18 | ||||
-rw-r--r-- | audio/decode/ad_mpg123.c | 2 | ||||
-rw-r--r-- | audio/decode/ad_spdif.c | 17 | ||||
-rw-r--r-- | audio/decode/dec_audio.c | 18 |
4 files changed, 34 insertions, 21 deletions
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c index 8177d9cde6..4997a66bc4 100644 --- a/audio/decode/ad_lavc.c +++ b/audio/decode/ad_lavc.c @@ -168,10 +168,16 @@ static int setup_format(sh_audio_t *sh_audio, else if (container_samplerate) samplerate = container_samplerate; - if (lavc_context->channels != sh_audio->channels || + struct mp_chmap lavc_chmap; + mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout); + // No channel layout or layout disagrees with channel count + if (lavc_chmap.num != lavc_context->channels) + mp_chmap_from_channels(&lavc_chmap, lavc_context->channels); + + if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) || samplerate != sh_audio->samplerate || sample_format != sh_audio->sample_format) { - sh_audio->channels = lavc_context->channels; + sh_audio->channels = lavc_chmap; sh_audio->samplerate = samplerate; sh_audio->sample_format = sample_format; sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8; @@ -227,8 +233,11 @@ static int init(sh_audio_t *sh_audio, const char *decoder) lavc_context->codec_type = AVMEDIA_TYPE_AUDIO; lavc_context->codec_id = lavc_codec->id; - if (opts->downmix) - lavc_context->request_channels = mpopts->audio_output_channels; + if (opts->downmix) { + lavc_context->request_channels = mpopts->audio_output_channels.num; + lavc_context->request_channel_layout = + mp_chmap_to_lavc(&mpopts->audio_output_channels); + } // Always try to set - option only exists for AC3 at the moment av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc, @@ -246,6 +255,7 @@ static int init(sh_audio_t *sh_audio, const char *decoder) lavc_context->codec_tag = sh_audio->format; lavc_context->sample_rate = sh_audio->samplerate; lavc_context->bit_rate = sh_audio->i_bps * 8; + lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels); if (sh_audio->wf) set_from_wf(lavc_context, sh_audio->wf); diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c index 999dc2fbba..45538f42f6 100644 --- a/audio/decode/ad_mpg123.c +++ b/audio/decode/ad_mpg123.c @@ -358,7 +358,7 @@ static int init(sh_audio_t *sh, const char *decoder) con->mean_count = 0; #endif con->vbr = (finfo.vbr != MPG123_CBR); - sh->channels = channels; + mp_chmap_from_channels(&sh->channels, channels); sh->samplerate = rate; /* Without external force, mpg123 will always choose signed encoding, * and non-16-bit only on builds that don't support it. diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c index ad735dde7d..a6f41932e9 100644 --- a/audio/decode/ad_spdif.c +++ b/audio/decode/ad_spdif.c @@ -148,19 +148,20 @@ static int init(sh_audio_t *sh, const char *decoder) } sh->ds->buffer_pos -= in_size; + int num_channels = 0; switch (lavf_ctx->streams[0]->codec->codec_id) { case AV_CODEC_ID_AAC: spdif_ctx->iec61937_packet_size = 16384; sh->sample_format = AF_FORMAT_IEC61937_LE; sh->samplerate = srate; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; break; case AV_CODEC_ID_AC3: spdif_ctx->iec61937_packet_size = 6144; sh->sample_format = AF_FORMAT_AC3_LE; sh->samplerate = srate; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; break; case AV_CODEC_ID_DTS: @@ -175,13 +176,13 @@ static int init(sh_audio_t *sh, const char *decoder) spdif_ctx->iec61937_packet_size = 32768; sh->sample_format = AF_FORMAT_IEC61937_LE; sh->samplerate = 192000; // DTS core require 48000 - sh->channels = 2*4; + num_channels = 2*4; sh->i_bps = bps; } else { spdif_ctx->iec61937_packet_size = 32768; sh->sample_format = AF_FORMAT_AC3_LE; sh->samplerate = srate; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; } break; @@ -189,26 +190,28 @@ static int init(sh_audio_t *sh, const char *decoder) spdif_ctx->iec61937_packet_size = 24576; sh->sample_format = AF_FORMAT_IEC61937_LE; sh->samplerate = 192000; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; break; case AV_CODEC_ID_MP3: spdif_ctx->iec61937_packet_size = 4608; sh->sample_format = AF_FORMAT_MPEG2; sh->samplerate = srate; - sh->channels = 2; + num_channels = 2; sh->i_bps = bps; break; case AV_CODEC_ID_TRUEHD: spdif_ctx->iec61937_packet_size = 61440; sh->sample_format = AF_FORMAT_IEC61937_LE; sh->samplerate = 192000; - sh->channels = 8; + num_channels = 8; sh->i_bps = bps; break; default: break; } + if (num_channels) + mp_chmap_from_channels(&sh->channels, num_channels); return 1; diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c index 11232f9271..999a96a10b 100644 --- a/audio/decode/dec_audio.c +++ b/audio/decode/dec_audio.c @@ -86,7 +86,7 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder) sh_audio->initialized = 1; - if (!sh_audio->channels || !sh_audio->samplerate) { + if (mp_chmap_is_empty(&sh_audio->channels) || !sh_audio->samplerate) { mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify " "audio format!\n"); uninit_audio(sh_audio); // free buffers @@ -94,7 +94,7 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder) } if (!sh_audio->o_bps) - sh_audio->o_bps = sh_audio->channels * sh_audio->samplerate + sh_audio->o_bps = sh_audio->channels.num * sh_audio->samplerate * sh_audio->samplesize; return 1; } @@ -160,14 +160,14 @@ int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders) sh_audio->gsh->decoder_desc); mp_msg(MSGT_DECAUDIO, MSGL_V, "AUDIO: %d Hz, %d ch, %s, %3.1f kbit/%3.2f%% (ratio: %d->%d)\n", - sh_audio->samplerate, sh_audio->channels, + sh_audio->samplerate, sh_audio->channels.num, af_fmt2str_short(sh_audio->sample_format), sh_audio->i_bps * 8 * 0.001, ((float) sh_audio->i_bps / sh_audio->o_bps) * 100.0, sh_audio->i_bps, sh_audio->o_bps); mp_msg(MSGT_IDENTIFY, MSGL_INFO, "ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n", - sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels); + sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels.num); } else { mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Failed to initialize an audio decoder for codec '%s'.\n", @@ -207,7 +207,7 @@ int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate, afs = af_new(sh_audio->opts); // input format: same as codec's output format: afs->input.rate = in_samplerate; - mp_audio_set_num_channels(&afs->input, sh_audio->channels); + mp_audio_set_channels(&afs->input, &sh_audio->channels); mp_audio_set_format(&afs->input, sh_audio->sample_format); // output format: same as ao driver's input format (if missing, fallback to input) @@ -259,7 +259,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len) // Decode more bytes if needed int old_samplerate = sh->samplerate; - int old_channels = sh->channels; + struct mp_chmap old_channels = sh->channels; int old_sample_format = sh->sample_format; while (sh->a_buffer_len < len) { unsigned char *buf = sh->a_buffer + sh->a_buffer_len; @@ -267,7 +267,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len) int maxlen = sh->a_buffer_size - sh->a_buffer_len; int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen); int format_change = sh->samplerate != old_samplerate - || sh->channels != old_channels + || !mp_chmap_equals(&sh->channels, &old_channels) || sh->sample_format != old_sample_format; if (ret <= 0 || format_change) { error = format_change ? -2 : -1; @@ -285,7 +285,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len) .rate = sh->samplerate, }; mp_audio_set_format(&filter_input, sh->sample_format); - mp_audio_set_num_channels(&filter_input, sh->channels); + mp_audio_set_channels(&filter_input, &sh->channels); struct mp_audio *filter_output = af_play(sh->afilter, &filter_input); if (!filter_output) @@ -314,7 +314,7 @@ int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen) // Indicates that a filter seems to be buffering large amounts of data int huge_filter_buffer = 0; // Decoded audio must be cut at boundaries of this many bytes - int unitsize = sh_audio->channels * sh_audio->samplesize * 16; + int unitsize = sh_audio->channels.num * sh_audio->samplesize * 16; /* Filter output size will be about filter_multiplier times input size. * If some filter buffers audio in big blocks this might only hold |