diff options
-rw-r--r-- | audio/decode/ad.h | 4 | ||||
-rw-r--r-- | audio/decode/ad_lavc.c | 161 | ||||
-rw-r--r-- | audio/decode/ad_mpg123.c | 393 | ||||
-rw-r--r-- | audio/decode/ad_spdif.c | 55 | ||||
-rw-r--r-- | audio/decode/dec_audio.c | 149 | ||||
-rw-r--r-- | audio/reorder_ch.c | 32 | ||||
-rw-r--r-- | audio/reorder_ch.h | 2 | ||||
-rw-r--r-- | demux/demux.c | 1 | ||||
-rw-r--r-- | demux/stheader.h | 9 | ||||
-rw-r--r-- | mpvcore/player/audio.c | 13 |
10 files changed, 324 insertions, 495 deletions
diff --git a/audio/decode/ad.h b/audio/decode/ad.h index 30e739d135..6c76e8dfd0 100644 --- a/audio/decode/ad.h +++ b/audio/decode/ad.h @@ -24,6 +24,7 @@ #include "demux/demux.h" #include "audio/format.h" +#include "audio/audio.h" struct mp_decoder_list; @@ -35,8 +36,7 @@ struct ad_functions { int (*init)(sh_audio_t *sh, const char *decoder); void (*uninit)(sh_audio_t *sh); int (*control)(sh_audio_t *sh, int cmd, void *arg); - int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen, - int maxlen); + int (*decode_audio)(sh_audio_t *sh, struct mp_audio *buffer, int maxlen); }; enum ad_ctrl { diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c index 1e63f0c3f2..c42c430850 100644 --- a/audio/decode/ad_lavc.c +++ b/audio/decode/ad_lavc.c @@ -36,25 +36,20 @@ #include "mpvcore/av_opts.h" #include "ad.h" -#include "audio/reorder_ch.h" #include "audio/fmt-conversion.h" -#include "compat/mpbswap.h" #include "compat/libav.h" struct priv { AVCodecContext *avctx; AVFrame *avframe; - uint8_t *output; - uint8_t *output_packed; // used by deplanarize to store packed audio samples - int output_left; - int unitsize; + struct mp_audio frame; bool force_channel_map; struct demux_packet *packet; }; static void uninit(sh_audio_t *sh); -static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen); +static int decode_new_packet(struct sh_audio *sh); #define OPT_BASE_STRUCT struct MPOpts @@ -150,22 +145,21 @@ static int preinit(sh_audio_t *sh) return 1; } -/* Prefer playing audio with the samplerate given in container data - * if available, but take number the number of channels and sample format - * from the codec, since if the codec isn't using the correct values for - * those everything breaks anyway. - */ -static int setup_format(sh_audio_t *sh_audio, - const AVCodecContext *lavc_context) +static int setup_format(sh_audio_t *sh_audio) { struct priv *priv = sh_audio->context; - int sample_format = - af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt)); - int samplerate = lavc_context->sample_rate; - // If not set, try container samplerate + AVCodecContext *lavc_context = priv->avctx; + + int sample_format = af_from_avformat(lavc_context->sample_fmt); + if (!sample_format) + return -1; + + int samplerate = lavc_context->sample_rate; if (!samplerate && sh_audio->wf) { + // If not set, try container samplerate. + // (Maybe this can't happen, and it's an artifact from the past.) samplerate = sh_audio->wf->nSamplesPerSec; - mp_tmsg(MSGT_DECAUDIO, MSGL_V, "ad_lavc: using container rate.\n"); + mp_tmsg(MSGT_DECAUDIO, MSGL_WARN, "ad_lavc: using container rate.\n"); } struct mp_chmap lavc_chmap; @@ -178,14 +172,9 @@ static int setup_format(sh_audio_t *sh_audio, lavc_chmap = sh_audio->channels; } - if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) || - samplerate != sh_audio->samplerate || - sample_format != sh_audio->sample_format) { - sh_audio->channels = lavc_chmap; - sh_audio->samplerate = samplerate; - sh_audio->sample_format = sample_format; - return 1; - } + sh_audio->channels = lavc_chmap; + sh_audio->samplerate = samplerate; + sh_audio->sample_format = sample_format; return 0; } @@ -285,15 +274,12 @@ static int init(sh_audio_t *sh_audio, const char *decoder) mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); - // Decode at least 1 byte: (to get header filled) - for (int tries = 0;;) { - int x = decode_audio(sh_audio, sh_audio->a_buffer, 1, - sh_audio->a_buffer_size); - if (x > 0) { - sh_audio->a_buffer_len = x; + // Decode at least 1 sample: (to get header filled) + for (int tries = 1; ; tries++) { + int x = decode_new_packet(sh_audio); + if (x >= 0 && ctx->frame.samples > 0) break; - } - if (++tries >= 5) { + if (tries >= 5) { mp_msg(MSGT_DECAUDIO, MSGL_ERR, "ad_lavc: initial decode failed\n"); uninit(sh_audio); @@ -305,12 +291,6 @@ static int init(sh_audio_t *sh_audio, const char *decoder) if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec; - int af_sample_fmt = - af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt)); - if (af_sample_fmt == AF_FORMAT_UNKNOWN) { - uninit(sh_audio); - return 0; - } return 1; } @@ -338,7 +318,7 @@ static int control(sh_audio_t *sh, int cmd, void *arg) switch (cmd) { case ADCTRL_RESYNC_STREAM: avcodec_flush_buffers(ctx->avctx); - ctx->output_left = 0; + ctx->frame.samples = 0; talloc_free(ctx->packet); ctx->packet = NULL; return CONTROL_TRUE; @@ -346,29 +326,13 @@ static int control(sh_audio_t *sh, int cmd, void *arg) return CONTROL_UNKNOWN; } -static av_always_inline void deplanarize(struct sh_audio *sh) -{ - struct priv *priv = sh->context; - - uint8_t **planes = priv->avframe->extended_data; - size_t bps = av_get_bytes_per_sample(priv->avctx->sample_fmt); - size_t nb_samples = priv->avframe->nb_samples; - size_t channels = priv->avctx->channels; - size_t size = bps * nb_samples * channels; - - if (talloc_get_size(priv->output_packed) != size) - priv->output_packed = - talloc_realloc_size(priv, priv->output_packed, size); - - reorder_to_packed(priv->output_packed, planes, bps, channels, nb_samples); - - priv->output = priv->output_packed; -} - static int decode_new_packet(struct sh_audio *sh) { struct priv *priv = sh->context; AVCodecContext *avctx = priv->avctx; + + priv->frame.samples = 0; + struct demux_packet *mpkt = priv->packet; if (!mpkt) mpkt = demux_read_packet(sh->gsh); @@ -384,7 +348,7 @@ static int decode_new_packet(struct sh_audio *sh) if (mpkt->pts != MP_NOPTS_VALUE) { sh->pts = mpkt->pts; - sh->pts_bytes = 0; + sh->pts_offset = 0; } int got_frame = 0; int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt); @@ -409,58 +373,39 @@ static int decode_new_packet(struct sh_audio *sh) } if (!got_frame) return 0; - uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) * - avctx->channels; - if (unitsize > 100000) - abort(); - priv->unitsize = unitsize; - uint64_t output_left = unitsize * priv->avframe->nb_samples; - if (output_left > 500000000) - abort(); - priv->output_left = output_left; - if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) { - deplanarize(sh); - } else { - priv->output = priv->avframe->data[0]; - } - mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", in_len, - priv->output_left); + + if (setup_format(sh) < 0) + return -1; + + priv->frame.samples = priv->avframe->nb_samples; + mp_audio_set_format(&priv->frame, sh->sample_format); + mp_audio_set_channels(&priv->frame, &sh->channels); + priv->frame.rate = sh->samplerate; + for (int n = 0; n < priv->frame.num_planes; n++) + priv->frame.planes[n] = priv->avframe->data[n]; + + mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d samples\n", in_len, + priv->frame.samples); return 0; } - -static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen, - int maxlen) +static int decode_audio(sh_audio_t *sh, struct mp_audio *buffer, int maxlen) { - struct priv *priv = sh_audio->context; - AVCodecContext *avctx = priv->avctx; + struct priv *priv = sh->context; - int len = -1; - while (len < minlen) { - if (!priv->output_left) { - if (decode_new_packet(sh_audio) < 0) - break; - continue; - } - if (setup_format(sh_audio, avctx)) - return len; - int size = (minlen - len + priv->unitsize - 1); - size -= size % priv->unitsize; - size = FFMIN(size, priv->output_left); - if (size > maxlen) - abort(); - memcpy(buf, priv->output, size); - priv->output += size; - priv->output_left -= size; - if (len < 0) - len = size; - else - len += size; - buf += size; - maxlen -= size; - sh_audio->pts_bytes += size; + if (!priv->frame.samples) { + if (decode_new_packet(sh) < 0) + return -1; } - return len; + + if (!mp_audio_config_equals(buffer, &priv->frame)) + return 0; + + buffer->samples = MPMIN(priv->frame.samples, maxlen); + mp_audio_copy(buffer, 0, &priv->frame, 0, buffer->samples); + mp_audio_skip_samples(&priv->frame, buffer->samples); + sh->pts_offset += buffer->samples; + return 0; } static void add_decoders(struct mp_decoder_list *list) diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c index 609e68f1c8..322f45826f 100644 --- a/audio/decode/ad_mpg123.c +++ b/audio/decode/ad_mpg123.c @@ -35,7 +35,9 @@ struct ad_mpg123_context { mpg123_handle *handle; - char new_format; + bool new_format; + int sample_size; + bool need_data; /* Running mean for bit rate, stream length estimation. */ float mean_rate; unsigned int mean_count; @@ -45,6 +47,17 @@ struct ad_mpg123_context { char vbr; }; +static void uninit(sh_audio_t *sh) +{ + struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context; + + mpg123_close(con->handle); + mpg123_delete(con->handle); + talloc_free(sh->context); + sh->context = NULL; + mpg123_exit(); +} + /* This initializes libmpg123 and prepares the handle, including funky * parameters. */ static int preinit(sh_audio_t *sh) @@ -58,7 +71,7 @@ static int preinit(sh_audio_t *sh) if (mpg123_init() != MPG123_OK) return 0; - sh->context = malloc(sizeof(struct ad_mpg123_context)); + sh->context = talloc_zero(NULL, struct ad_mpg123_context); con = sh->context; /* Auto-choice of optimized decoder (first argument NULL). */ con->handle = mpg123_new(NULL, &err); @@ -92,7 +105,12 @@ static int preinit(sh_audio_t *sh) /* Prevent funky automatic resampling. * This way, we can be sure that one frame will never produce - * more than 1152 stereo samples. */ + * more than 1152 stereo samples. + * Background: + * Going to decode directly to the output buffer. It is important to have + * MPG123_AUTO_RESAMPLE disabled for the buffer size being an all-time + * limit. + * We need at least 1152 samples. dec_audio.c normally guarantees this. */ mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.); return 1; @@ -105,77 +123,21 @@ static int preinit(sh_audio_t *sh) mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n", mpg123_strerror(con->handle)); - if (con->handle) - mpg123_delete(con->handle); - mpg123_exit(); - free(sh->context); - sh->context = NULL; + uninit(sh); return 0; } -/* Compute bitrate from frame size. */ -static int compute_bitrate(struct mpg123_frameinfo *i) -{ - static const int samples_per_frame[4][4] = { - {-1, 384, 1152, 1152}, /* MPEG 1 */ - {-1, 384, 1152, 576}, /* MPEG 2 */ - {-1, 384, 1152, 576}, /* MPEG 2.5 */ - {-1, -1, -1, -1}, /* Unknown */ - }; - return (int) ((i->framesize + 4) * 8 * i->rate * 0.001 / - samples_per_frame[i->version][i->layer] + 0.5); -} - -/* Opted against the header printout from old mp3lib, too much - * irrelevant info. This is modelled after the mpg123 app's - * standard output line. - * If more verbosity is demanded, one can add more detail and - * also throw in ID3v2 info which libmpg123 collects anyway. */ -static void print_header_compact(struct mpg123_frameinfo *i) -{ - static const char *smodes[5] = { - "stereo", "joint-stereo", "dual-channel", "mono", "invalid" - }; - static const char *layers[4] = { - "Unknown", "I", "II", "III" - }; - static const char *versions[4] = { - "1.0", "2.0", "2.5", "x.x" - }; - - mp_msg(MSGT_DECAUDIO, MSGL_V, "MPEG %s layer %s, ", - versions[i->version], layers[i->layer]); - switch (i->vbr) { - case MPG123_CBR: - if (i->bitrate) - mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s", i->bitrate); - else - mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s (free format)", - compute_bitrate(i)); - break; - case MPG123_VBR: - mp_msg(MSGT_DECAUDIO, MSGL_V, "VBR"); - break; - case MPG123_ABR: - mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s ABR", i->abr_rate); - break; - default: - mp_msg(MSGT_DECAUDIO, MSGL_V, "???"); - } - mp_msg(MSGT_DECAUDIO, MSGL_V, ", %ld Hz %s\n", i->rate, - smodes[i->mode]); -} - /* libmpg123 has a new format ready; query and store, return return value of mpg123_getformat() */ -static int set_format(sh_audio_t *sh, struct ad_mpg123_context *con) +static int set_format(sh_audio_t *sh) { + struct ad_mpg123_context *con = sh->context; int ret; long rate; int channels; int encoding; ret = mpg123_getformat(con->handle, &rate, &channels, &encoding); - if(ret == MPG123_OK) { + if (ret == MPG123_OK) { mp_chmap_from_channels(&sh->channels, channels); sh->samplerate = rate; /* Without external force, mpg123 will always choose signed encoding, @@ -189,13 +151,10 @@ static int set_format(sh_audio_t *sh, struct ad_mpg123_context *con) case MPG123_ENC_SIGNED_16: sh->sample_format = AF_FORMAT_S16_NE; break; - /* To stay compatible with the oldest libmpg123 headers, do not rely - * on float and 32 bit encoding symbols being defined. - * Those formats came later */ - case 0x1180: /* MPG123_ENC_SIGNED_32 */ + case MPG123_ENC_SIGNED_32: sh->sample_format = AF_FORMAT_S32_NE; break; - case 0x200: /* MPG123_ENC_FLOAT_32 */ + case MPG123_ENC_FLOAT_32: sh->sample_format = AF_FORMAT_FLOAT_NE; break; default: @@ -204,135 +163,38 @@ static int set_format(sh_audio_t *sh, struct ad_mpg123_context *con) "Bad encoding from mpg123: %i.\n", encoding); return MPG123_ERR; } - /* Going to decode directly to MPlayer's memory. It is important - * to have MPG123_AUTO_RESAMPLE disabled for the buffer size - * being an all-time limit. */ - sh->audio_out_minsize = 1152 * 2 * (af_fmt2bits(sh->sample_format) / 8); + con->sample_size = channels * (af_fmt2bits(sh->sample_format) / 8); con->new_format = 0; } return ret; } -/* This tries to extract a requested amount of decoded data. - * Even when you request 0 bytes, it will feed enough input so that - * the decoder _could_ have delivered something. - * Returns byte count >= 0, -1 on error. - * - * Thoughts on exact pts keeping: - * We have to assume that MPEG frames are cut in pieces by packet boundaries. - * Also, it might be possible that the first packet does not contain enough - * data to ensure initial stream sync... or re-sync on erroneous streams. - * So we need something robust to relate the decoded byte count to the correct - * time stamp. This is tricky, though. From the outside, you cannot tell if, - * after having fed two packets until the first output arrives, one should - * start counting from the first packet's pts or the second packet's. - * So, let's just count from the last fed package's pts. If the packets are - * exactly cut to MPEG frames, this will cause one frame mismatch in the - * beginning (when mpg123 peeks ahead for the following header), but will - * be corrected with the third frame already. One might add special code to - * not increment the base pts past the first packet's after a resync before - * the first decoded bytes arrived. */ -static int decode_a_bit(sh_audio_t *sh, unsigned char *buf, int count) +static int feed_new_packet(sh_audio_t *sh) { - int ret = MPG123_OK; - int got = 0; struct ad_mpg123_context *con = sh->context; + int ret; - /* There will be one MPG123_NEW_FORMAT message on first open. - * This will be handled in init(). */ - do { - size_t got_now = 0; - /* Fetch new format now, after old data has been used. */ - if(con->new_format) - ret = set_format(sh, con); - - /* Feed the decoder. This will only fire from the second round on. */ - if (ret == MPG123_NEED_MORE) { - /* Feed more input data. */ - struct demux_packet *pkt = demux_read_packet(sh->gsh); - if (!pkt) - break; /* Apparently that's it. EOF. */ - - /* Next bytes from that presentation time. */ - if (pkt->pts != MP_NOPTS_VALUE) { - sh->pts = pkt->pts; - sh->pts_bytes = 0; - } - - /* Have to use mpg123_feed() to avoid decoding here. */ - ret = mpg123_feed(con->handle, pkt->buffer, pkt->len); - talloc_free(pkt); - if (ret == MPG123_ERR) - break; - - /* Indication of format change is possible here (from mpg123_decode()). */ - if(ret == MPG123_NEW_FORMAT) { - con->new_format = 1; - if(got) - break; /* Do not switch format during a chunk. */ - - ret = set_format(sh, con); - } - } - /* Theoretically, mpg123 could return MPG123_DONE, so be prepared. - * Should not happen in our usage, but it is a valid return code. */ - else if (ret == MPG123_ERR || ret == MPG123_DONE) - break; - - /* Try to decode a bit. This is the return value that counts - * for the loop condition. */ - if (!buf) { /* fake call just for feeding to get format */ - ret = set_format(sh, con); - } else { /* This is the decoding. One frame at a time. */ - ret = mpg123_replace_buffer(con->handle, buf, count); - if (ret == MPG123_OK) - ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now); - } - - got += got_now; - sh->pts_bytes += got_now; - - /* Indication of format change should happen here. */ - if(ret == MPG123_NEW_FORMAT) { - con->new_format = 1; - if(got) - break; /* Do not switch format during a chunk. */ - - ret = set_format(sh, con); - } - - } while (ret == MPG123_NEED_MORE || (got == 0 && count != 0)); + struct demux_packet *pkt = demux_read_packet(sh->gsh); + if (!pkt) + return -1; /* EOF. */ - if (ret == MPG123_ERR) { - mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 decoding failed: %s\n", - mpg123_strerror(con->handle)); + /* Next bytes from that presentation time. */ + if (pkt->pts != MP_NOPTS_VALUE) { + sh->pts = pkt->pts; + sh->pts_offset = 0; } - return got; -} + /* Have to use mpg123_feed() to avoid decoding here. */ + ret = mpg123_feed(con->handle, pkt->buffer, pkt->len); + talloc_free(pkt); -/* Close, reopen stream. Feed data until we know the format of the stream. - * 1 on success, 0 on error */ -static int reopen_stream(sh_audio_t *sh) -{ - struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context; + if (ret == MPG123_ERR) + return -1; - mpg123_close(con->handle); - /* No resetting of the context: - * We do not want to loose the mean bitrate data. */ - - /* Open and make sure we have fed enough data to get stream properties. */ - if (MPG123_OK == mpg123_open_feed(con->handle) && - /* Feed data until mpg123 is ready (has found stream beginning). */ - !decode_a_bit(sh, NULL, 0) && - set_format(sh, con) == MPG123_OK) { /* format setting again just for return value */ - return 1; - } else { - mp_msg(MSGT_DECAUDIO, MSGL_ERR, - "mpg123 failed to reopen stream: %s\n", - mpg123_strerror(con->handle)); - return 0; - } + if (ret == MPG123_NEW_FORMAT) + con->new_format = 1; + + return 0; } /* Now we really start accessing some data and determining file format. @@ -341,49 +203,59 @@ static int reopen_stream(sh_audio_t *sh) * erros in other places simply cannot occur. */ static int init(sh_audio_t *sh, const char *decoder) { - mpg123_id3v2 *v2; - struct mpg123_frameinfo finfo; struct ad_mpg123_context *con = sh->context; + int ret; - con->new_format = 0; - if (reopen_stream(sh) && - /* Get MPEG header info. */ - MPG123_OK == mpg123_info(con->handle, &finfo) && - /* Since we queried format, mpg123 should have read past ID3v2 tags. - * We need to decide if printing of UTF-8 encoded text info is wanted. */ - MPG123_OK == mpg123_id3(con->handle, NULL, &v2)) { - /* If we are here, we passed all hurdles. Yay! Extract the info. */ - print_header_compact(&finfo); - /* Do we want to print out the UTF-8 Id3v2 info? - if (v2) - print_id3v2(v2); */ - - /* Have kb/s, want B/s - * For VBR, the first frame will be a bad estimate. */ - sh->i_bps = (finfo.bitrate ? finfo.bitrate : compute_bitrate(&finfo)) - * 1000 / 8; - con->delay = 1; - con->mean_rate = 0.; - con->mean_count = 0; - con->vbr = (finfo.vbr != MPG123_CBR); + ret = mpg123_open_feed(con->handle); + if (ret != MPG123_OK) + goto fail; + + for (int n = 0; ; n++) { + if (feed_new_packet(sh) < 0) { + ret = MPG123_NEED_MORE; + goto fail; + } + size_t got_now = 0; + ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now); + if (ret == MPG123_OK || ret == MPG123_NEW_FORMAT) { + ret = set_format(sh); + if (ret == MPG123_OK) + break; + } + if (ret != MPG123_NEED_MORE) + goto fail; + // max. 16 retries (randomly chosen number) + if (n > 16) { + ret = MPG123_NEED_MORE; + goto fail; + } + } + + return 1; - return 1; +fail: + if (ret == MPG123_NEED_MORE) { + mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Could not find mp3 stream.\n"); } else { mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 init error: %s\n", mpg123_strerror(con->handle)); - return 0; } + + uninit(sh); + return 0; } -static void uninit(sh_audio_t *sh) +/* Compute bitrate from frame size. */ +static int compute_bitrate(struct mpg123_frameinfo *i) { - struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context; - - mpg123_close(con->handle); - mpg123_delete(con->handle); - free(sh->context); - sh->context = NULL; - mpg123_exit(); + static const int samples_per_frame[4][4] = { + {-1, 384, 1152, 1152}, /* MPEG 1 */ + {-1, 384, 1152, 576}, /* MPEG 2 */ + {-1, 384, 1152, 576}, /* MPEG 2.5 */ + {-1, -1, -1, -1}, /* Unknown */ + }; + return (int) ((i->framesize + 4) * 8 * i->rate * 0.001 / + samples_per_frame[i->version][i->layer] + 0.5); } /* Update mean bitrate. This could be dropped if accurate time display @@ -391,9 +263,12 @@ static void uninit(sh_audio_t *sh) static void update_info(sh_audio_t *sh) { struct ad_mpg123_context *con = sh->context; - if (con->vbr && --con->delay < 1) { - struct mpg123_frameinfo finfo; - if (MPG123_OK == mpg123_info(con->handle, &finfo)) { + struct mpg123_frameinfo finfo; + if (mpg123_info(con->handle, &finfo) != MPG123_OK) + return; + + if (finfo.vbr != MPG123_CBR) { + if (--con->delay < 1) { if (++con->mean_count > ((unsigned int) -1) / 2) con->mean_count = ((unsigned int) -1) / 4; @@ -404,42 +279,80 @@ static void update_info(sh_audio_t *sh) con->delay = 10; } + } else { + sh->i_bps = (finfo.bitrate ? finfo.bitrate : compute_bitrate(&finfo)) + * 1000 / 8; + con->delay = 1; + con->mean_rate = 0.; + con->mean_count = 0; } } -static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen, - int maxlen) +static int decode_audio(sh_audio_t *sh, struct mp_audio *buffer, int maxlen) { - int bytes; + struct ad_mpg123_context *con = sh->context; + void *buf = buffer->planes[0]; + int ret; - bytes = decode_a_bit(sh, buf, maxlen); - /* This EOF is ignored, apparently, until input data is exhausted. */ - if (bytes == 0) - return -1; /* EOF */ + if (con->new_format) { + ret = set_format(sh); + if (ret == MPG123_OK) { + return 0; // let caller handle format change + } else if (ret == MPG123_NEED_MORE) { + con->need_data = true; + } else { + goto mpg123_fail; + } + } + + if (con->need_data) { + if (feed_new_packet(sh) < 0) + return -1; + } + + size_t got_now = 0; + ret = mpg123_replace_buffer(con->handle, buf, maxlen * con->sample_size); + if (ret != MPG123_OK) + goto mpg123_fail; + + ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now); + + int got_samples = got_now / con->sample_size; + buffer->samples += got_samples; + sh->pts_offset += got_samples; + + if (ret == MPG123_NEW_FORMAT) { + con->new_format = true; + } else if (ret == MPG123_NEED_MORE) { + con->need_data = true; + } else if (ret != MPG123_OK && ret != MPG123_DONE) { + goto mpg123_fail; + } update_info(sh); - return bytes; + return 0; + +mpg123_fail: + mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 decoding error: %s\n", + mpg123_strerror(con->handle)); + return -1; } static int control(sh_audio_t *sh, int cmd, void *arg) { + struct ad_mpg123_context *con = sh->context; + switch (cmd) { case ADCTRL_RESYNC_STREAM: - /* Close/reopen the stream for mpg123 to make sure it doesn't - * think that it still knows the exact stream position. - * Otherwise, we would have funny effects from the gapless code. - * Oh, and it helps to minimize artifacts from jumping in the stream. */ - if (reopen_stream(sh)) { - update_info(sh); - return CONTROL_TRUE; - } else { - /* MPlayer ignores this case! It just keeps on decoding. - * So we have to make sure resync never fails ... */ + mpg123_close(con->handle); + + if (mpg123_open_feed(con->handle) != MPG123_OK) { mp_msg(MSGT_DECAUDIO, MSGL_ERR, - "mpg123 cannot reopen stream for resync.\n"); + "mpg123 failed to reopen stream: %s\n", + mpg123_strerror(con->handle)); return CONTROL_FALSE; } - break; + return CONTROL_TRUE; } return CONTROL_UNKNOWN; } diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c index f03041d6a6..a233286c19 100644 --- a/audio/decode/ad_spdif.c +++ b/audio/decode/ad_spdif.c @@ -19,6 +19,7 @@ */ #include <string.h> +#include <assert.h> #include <libavformat/avformat.h> #include <libavcodec/avcodec.h> @@ -184,37 +185,43 @@ fail: return 0; } -static int decode_audio(sh_audio_t *sh, unsigned char *buf, - int minlen, int maxlen) +static int decode_audio(sh_audio_t *sh, struct mp_audio *buffer, int maxlen) { struct spdifContext *spdif_ctx = sh->context; AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx; + int sstride = 2 * sh->channels.num; + assert(sstride == buffer->sstride); + + if (maxlen < spdif_ctx->iec61937_packet_size) + return 0; + spdif_ctx->out_buffer_len = 0; spdif_ctx->out_buffer_size = maxlen; - spdif_ctx->out_buffer = buf; - while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen - && spdif_ctx->out_buffer_len < minlen) { - struct demux_packet *mpkt = demux_read_packet(sh->gsh); - if (!mpkt) - break; - AVPacket pkt; - mp_set_av_packet(&pkt, mpkt); - pkt.pts = pkt.dts = 0; - mp_msg(MSGT_DECAUDIO, MSGL_V, "spdif packet, size=%d\n", pkt.size); - if (mpkt->pts != MP_NOPTS_VALUE) { - sh->pts = mpkt->pts; - sh->pts_bytes = 0; - } - int out_len = spdif_ctx->out_buffer_len; - int ret = av_write_frame(lavf_ctx, &pkt); - avio_flush(lavf_ctx->pb); - sh->pts_bytes += spdif_ctx->out_buffer_len - out_len; - talloc_free(mpkt); - if (ret < 0) - break; + spdif_ctx->out_buffer = buffer->planes[0]; + + struct demux_packet *mpkt = demux_read_packet(sh->gsh); + if (!mpkt) + return 0; + + AVPacket pkt; + mp_set_av_packet(&pkt, mpkt); + pkt.pts = pkt.dts = 0; + mp_msg(MSGT_DECAUDIO, MSGL_V, "spdif packet, size=%d\n", pkt.size); + if (mpkt->pts != MP_NOPTS_VALUE) { + sh->pts = mpkt->pts; + sh->pts_offset = 0; } - return spdif_ctx->out_buffer_len; + int out_len = spdif_ctx->out_buffer_len; + int ret = av_write_frame(lavf_ctx, &pkt); + avio_flush(lavf_ctx->pb); + sh->pts_offset += (spdif_ctx->out_buffer_len - out_len) / sstride; + talloc_free(mpkt); + if (ret < 0) + return -1; + + buffer->samples = spdif_ctx->out_buffer_len / sstride; + return 0; } static int control(sh_audio_t *sh, int cmd, void *arg) diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c index 6c4af89ef9..19b5d8bdeb 100644 --- a/audio/decode/dec_audio.c +++ b/audio/decode/dec_audio.c @@ -38,6 +38,7 @@ #include "dec_audio.h" #include "ad.h" #include "audio/format.h" +#include "audio/audio.h" #include "audio/audio_buffer.h" #include "audio/filter/af.h" @@ -55,31 +56,29 @@ static const struct ad_functions * const ad_drivers[] = { NULL }; +// At least ad_mpg123 needs to be able to decode this many samples at once +#define DECODE_MAX_UNIT 1152 + +// At least 8192 samples, plus hack for ad_mpg123 +#define DECODE_BUFFER_SAMPLES (8192 + DECODE_MAX_UNIT) + +// Drop audio buffer and reinit it (after format change) +static void reinit_audio_buffer(sh_audio_t *sh) +{ + mp_audio_buffer_reinit_fmt(sh->decode_buffer, sh->sample_format, + &sh->channels, sh->samplerate); + mp_audio_buffer_preallocate_min(sh->decode_buffer, DECODE_BUFFER_SAMPLES); +} + static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder) { assert(!sh_audio->initialized); resync_audio_stream(sh_audio); - sh_audio->sample_format = AF_FORMAT_FLOAT_NE; - sh_audio->audio_out_minsize = 8192; // default, preinit() may change it if (!sh_audio->ad_driver->preinit(sh_audio)) { mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder preinit failed.\n"); return 0; } - const int base_size = 65536; - // At least 64 KiB plus rounding up to next decodable unit size - sh_audio->a_buffer_size = base_size + sh_audio->audio_out_minsize; - - mp_tmsg(MSGT_DECAUDIO, MSGL_V, - "dec_audio: Allocating %d + %d = %d bytes for output buffer.\n", - sh_audio->audio_out_minsize, base_size, - sh_audio->a_buffer_size); - - sh_audio->a_buffer = av_mallocz(sh_audio->a_buffer_size); - if (!sh_audio->a_buffer) - abort(); - sh_audio->a_buffer_len = 0; - if (!sh_audio->ad_driver->init(sh_audio, decoder)) { mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Audio decoder init failed.\n"); uninit_audio(sh_audio); // free buffers @@ -88,13 +87,18 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder) sh_audio-> |