static int fill_queue(struct af_instance *af, struct mp_audio *data, int offset)
{
af_scaletempo_t *s = af->priv;
int bytes_in = mp_audio_psize(data) - offset;
int offset_unchanged = offset;
if (s->bytes_to_slide > 0) {
if (s->bytes_to_slide < s->bytes_queued) {
int bytes_move = s->bytes_queued - s->bytes_to_slide;
memmove(s->buf_queue, s->buf_queue + s->bytes_to_slide, bytes_move);
s->bytes_to_slide = 0;
s->bytes_queued = bytes_move;
} else {
int bytes_skip;
s->bytes_to_slide -= s->bytes_queued;
bytes_skip = MPMIN(s->bytes_to_slide, bytes_in);
s->bytes_queued = 0;
s->bytes_to_slide -= bytes_skip;
offset += bytes_skip;
bytes_in -= bytes_skip;
}
}
if (bytes_in > 0) {
int bytes_copy = MPMIN(s->bytes_queue - s->bytes_queued, bytes_in);
assert(bytes_copy >= 0);
memcpy(s->buf_queue + s->bytes_queued,
(int8_t *)data->planes[0] + offset, bytes_copy);
s->bytes_queued += bytes_copy;
offset += bytes_copy;
}
return offset - offset_unchanged;
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_pan_t* s = af->priv;
switch(cmd) {
case AF_CONTROL_REINIT:
// Sanity check
if(!arg) return AF_ERROR;
af->data->rate = ((struct mp_audio*)arg)->rate;
mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
set_channels(af->data, s->nch ? s->nch: ((struct mp_audio*)arg)->nch);
if((af->data->format != ((struct mp_audio*)arg)->format) ||
(af->data->bps != ((struct mp_audio*)arg)->bps)) {
mp_audio_set_format((struct mp_audio*)arg, af->data->format);
return AF_FALSE;
}
return AF_OK;
case AF_CONTROL_SET_PAN_LEVEL: {
int i;
int ch = ((af_control_ext_t*)arg)->ch;
float* level = ((af_control_ext_t*)arg)->arg;
if (ch >= AF_NCH)
return AF_FALSE;
for(i=0; i<AF_NCH; i++)
s->level[ch][i] = level[i];
return AF_OK;
}
case AF_CONTROL_SET_PAN_NOUT:
// Reinit must be called after this function has been called
// Sanity check
if(((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH) {
MP_ERR(af, "The number of output channels must be"
" between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]);
return AF_ERROR;
}
s->nch=((int*)arg)[0];
return AF_OK;
case AF_CONTROL_SET_PAN_BALANCE: {
float val = *(float*)arg;
if (s->nch)
return AF_ERROR;
if (af->data->nch >= 2) {
s->level[0][0] = MPMIN(1.f, 1.f - val);
s->level[0][1] = MPMAX(0.f, val);
s->level[1][0] = MPMAX(0.f, -val);
s->level[1][1] = MPMIN(1.f, 1.f + val);
}
return AF_OK;
}
case AF_CONTROL_GET_PAN_BALANCE:
if (s->nch)
return AF_ERROR;
*(float*)arg = s->level[0][1] - s->level[1][0];
return AF_OK;
}
return AF_UNKNOWN;
}
// This is called both during init and at runtime.
static int resize_cache(struct priv *s, int64_t size)
{
int64_t min_size = FILL_LIMIT * 4;
int64_t max_size = ((size_t)-1) / 4;
int64_t buffer_size = MPMIN(MPMAX(size, min_size), max_size);
unsigned char *buffer = malloc(buffer_size);
struct byte_meta *bm = calloc(buffer_size / BYTE_META_CHUNK_SIZE + 2,
sizeof(struct byte_meta));
if (!buffer || !bm) {
free(buffer);
free(bm);
return STREAM_ERROR;
}
if (s->buffer) {
// Copy & free the old ringbuffer data.
// If the buffer is too small, prefer to copy these regions:
// 1. Data starting from read_filepos, until cache end
size_t read_1 = read_buffer(s, buffer, buffer_size, s->read_filepos);
// 2. then data from before read_filepos until cache start
// (this one needs to be copied to the end of the ringbuffer)
size_t read_2 = 0;
if (s->min_filepos < s->read_filepos) {
size_t copy_len = buffer_size - read_1;
copy_len = MPMIN(copy_len, s->read_filepos - s->min_filepos);
assert(copy_len + read_1 <= buffer_size);
read_2 = read_buffer(s, buffer + buffer_size - copy_len, copy_len,
s->read_filepos - copy_len);
// This shouldn't happen, unless copy_len was computed incorrectly.
assert(read_2 == copy_len);
}
// Set it up such that read_1 is at buffer pos 0, and read_2 wraps
// around below it, so that it is located at the end of the buffer.
s->min_filepos = s->read_filepos - read_2;
s->max_filepos = s->read_filepos + read_1;
s->offset = s->max_filepos - read_1;
} else {
cache_drop_contents(s);
}
free(s->buffer);
free(s->bm);
s->buffer_size = buffer_size;
s->back_size = buffer_size / 2;
s->buffer = buffer;
s->bm = bm;
s->idle = false;
s->eof = false;
//make sure that we won't wait from cache_fill
//more data than it is allowed to fill
if (s->seek_limit > s->buffer_size - FILL_LIMIT)
s->seek_limit = s->buffer_size - FILL_LIMIT;
return STREAM_OK;
}
static void fill_rect(struct vo *vo, GC gc, int x0, int y0, int x1, int y1)
{
struct vo_x11_state *x11 = vo->x11;
x0 = MPMAX(x0, 0);
y0 = MPMAX(y0, 0);
x1 = MPMIN(x1, vo->dwidth);
y1 = MPMIN(y1, vo->dheight);
if (x11->window && gc && x1 > x0 && y1 > y0)
XFillRectangle(x11->display, x11->window, gc, x0, y0, x1 - x0, y1 - y0);
}
// Copy at most dst_size from the cache at the given absolute file position pos.
// Return number of bytes that could actually be read.
// Does not advance the file position, or change anything else.
// Can be called from anywhere, as long as the mutex is held.
static size_t read_buffer(struct priv *s, unsigned char *dst,
size_t dst_size, int64_t pos)
{
size_t read = 0;
while (read < dst_size) {
if (pos >= s->max_filepos || pos < s->min_filepos)
break;
int64_t newb = s->max_filepos - pos; // new bytes in the buffer
int64_t bpos = pos - s->offset; // file pos to buffer memory pos
if (bpos < 0) {
bpos += s->buffer_size;
} else if (bpos >= s->buffer_size) {
bpos -= s->buffer_size;
}
if (newb > s->buffer_size - bpos)
newb = s->buffer_size - bpos; // handle wrap...
newb = MPMIN(newb, dst_size - read);
assert(newb >= 0 && read + newb <= dst_size);
assert(bpos >= 0 && bpos + newb <= s->buffer_size);
memcpy(&dst[read], &s->buffer[bpos], newb);
read += newb;
pos += newb;
}
return read;
}
void ca_get_active_chmap(struct ao *ao, AudioDeviceID device, int channel_count,
struct mp_chmap *out_map)
{
// Apparently, we have to guess by looking back at the supported layouts,
// and I haven't found a property that retrieves the actual currently
// active channel layout.
struct mp_chmap_sel chmap_sel = {0};
ca_retrieve_layouts(ao, &chmap_sel, device);
// Use any exact match.
for (int n = 0; n < chmap_sel.num_chmaps; n++) {
if (chmap_sel.chmaps[n].num == channel_count) {
MP_VERBOSE(ao, "mismatching channels - fallback #%d\n", n);
*out_map = chmap_sel.chmaps[n];
return;
}
}
// Fall back to stereo or mono, and fill the rest with silence. (We don't
// know what the device expects. We could use a larger default layout here,
// but let's not.)
mp_chmap_from_channels(out_map, MPMIN(2, channel_count));
out_map->num = channel_count;
for (int n = 2; n < out_map->num; n++)
out_map->speaker[n] = MP_SPEAKER_ID_NA;
MP_WARN(ao, "mismatching channels - falling back to %s\n",
mp_chmap_to_str(out_map));
}
static int archive_entry_seek(stream_t *s, int64_t newpos)
{
struct priv *p = s->priv;
if (!p->mpa)
return -1;
if (archive_seek_data(p->mpa->arch, newpos, SEEK_SET) >= 0)
return 1;
// libarchive can't seek in most formats.
if (newpos < s->pos) {
// Hack seeking backwards into working by reopening the archive and
// starting over.
MP_VERBOSE(s, "trying to reopen archive for performing seek\n");
if (reopen_archive(s) < STREAM_OK)
return -1;
s->pos = 0;
}
if (newpos > s->pos) {
// For seeking forwards, just keep reading data (there's no libarchive
// skip function either).
char buffer[4096];
while (newpos > s->pos) {
if (mp_cancel_test(s->cancel))
return -1;
int size = MPMIN(newpos - s->pos, sizeof(buffer));
int r = archive_read_data(p->mpa->arch, buffer, size);
if (r < 0) {
MP_ERR(s, "%s\n", archive_error_string(p->mpa->arch));
return -1;
}
s->pos += r;
}
}
return 1;
}
static void mp_msg_av_log_callback(void *ptr, int level, const char *fmt,
va_list vl)
{
AVClass *avc = ptr ? *(AVClass **)ptr : NULL;
int mp_level = av_log_level_to_mp_level(level);
// Note: mp_log is thread-safe, but destruction of the log instances is not.
pthread_mutex_lock(&log_lock);
if (!log_mpv_instance) {
pthread_mutex_unlock(&log_lock);
// Fallback to stderr
vfprintf(stderr, fmt, vl);
return;
}
struct mp_log *log = get_av_log(ptr);
if (mp_msg_test(log, mp_level)) {
char buffer[4096] = "";
int pos = 0;
const char *prefix = avc ? avc->item_name(ptr) : NULL;
if (log_print_prefix && prefix)
pos = snprintf(buffer, sizeof(buffer), "%s: ", prefix);
log_print_prefix = fmt[strlen(fmt) - 1] == '\n';
pos = MPMIN(MPMAX(pos, 0), sizeof(buffer));
vsnprintf(buffer + pos, sizeof(buffer) - pos, fmt, vl);
mp_msg(log, mp_level, "%s", buffer);
}
pthread_mutex_unlock(&log_lock);
}
static int filter_out(struct af_instance *af)
{
struct priv *p = af->priv;
while (rubberband_available(p->rubber) <= 0) {
const float *dummy[MP_NUM_CHANNELS] = {0};
const float **in_data = dummy;
size_t in_samples = 0;
if (p->pending) {
if (!p->pending->samples)
break;
// recover from previous EOF
if (p->needs_reset) {
rubberband_reset(p->rubber);
p->rubber_delay = 0;
}
p->needs_reset = false;
size_t needs = rubberband_get_samples_required(p->rubber);
in_data = (void *)&p->pending->planes;
in_samples = MPMIN(p->pending->samples, needs);
}
if (p->needs_reset)
break; // previous EOF
p->needs_reset = !p->pending; // EOF
rubberband_process(p->rubber, in_data, in_samples, p->needs_reset);
p->rubber_delay += in_samples;
if (!p->pending)
break;
mp_audio_skip_samples(p->pending, in_samples);
}
int out_samples = rubberband_available(p->rubber);
if (out_samples > 0) {
struct mp_audio *out =
mp_audio_pool_get(af->out_pool, af->data, out_samples);
if (!out)
return -1;
if (p->pending)
mp_audio_copy_config(out, p->pending);
float **out_data = (void *)&out->planes;
out->samples = rubberband_retrieve(p->rubber, out_data, out->samples);
p->rubber_delay -= out->samples * p->speed;
af_add_output_frame(af, out);
}
int delay_samples = p->rubber_delay;
if (p->pending)
delay_samples += p->pending->samples;
af->delay = delay_samples / (af->data->rate * p->speed);
return 0;
}
// Filter len bytes of input, put result into outbuf.
static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
int len)
{
int error = 0;
struct mp_audio config;
mp_audio_buffer_get_format(da->decode_buffer, &config);
while (mp_audio_buffer_samples(da->decode_buffer) < len) {
int maxlen = mp_audio_buffer_get_write_available(da->decode_buffer);
if (maxlen < DECODE_MAX_UNIT)
break;
struct mp_audio buffer;
mp_audio_buffer_get_write_buffer(da->decode_buffer, maxlen, &buffer);
buffer.samples = 0;
error = da->ad_driver->decode_audio(da, &buffer, maxlen);
if (error < 0)
break;
// Commit the data just read as valid data
mp_audio_buffer_finish_write(da->decode_buffer, buffer.samples);
// Format change
if (!mp_audio_config_equals(&da->decoded, &config)) {
// If there are still samples left in the buffer, let them drain
// first, and don't signal a format change to the caller yet.
if (mp_audio_buffer_samples(da->decode_buffer) > 0)
break;
error = -2;
break;
}
}
// Filter
struct mp_audio filter_data;
mp_audio_buffer_peek(da->decode_buffer, &filter_data);
filter_data.rate = da->afilter->input.rate; // due to playback speed change
len = MPMIN(filter_data.samples, len);
filter_data.samples = len;
bool eof = filter_data.samples == 0 && error < 0;
if (af_filter(da->afilter, &filter_data, eof ? AF_FILTER_FLAG_EOF : 0) < 0)
return -1;
mp_audio_buffer_append(outbuf, &filter_data);
if (eof && filter_data.samples > 0)
error = 0; // don't end playback yet
// remove processed data from decoder buffer:
mp_audio_buffer_skip(da->decode_buffer, len);
// Assume the filter chain is drained from old data at this point.
// (If not, the remaining old data is discarded.)
if (error == -2) {
if (!reinit_audio_buffer(da))
error = -1; // switch to invalid format
}
return error;
}
static int fill_buffer(stream_t *s, char *buffer, int max_len)
{
struct priv *p = s->priv;
if (s->pos < 0)
return -1;
if (s->pos >= p->max_size) {
if (stream_seek(p->original, s->pos) < 1)
return -1;
return stream_read(p->original, buffer, max_len);
}
// Size of file changes -> invalidate last block
if (s->pos >= p->size - BLOCK_SIZE) {
int64_t new_size = -1;
stream_control(s, STREAM_CTRL_GET_SIZE, &new_size);
if (new_size != p->size)
set_bit(p, BLOCK_ALIGN(p->size), 0);
p->size = MPMIN(p->max_size, new_size);
}
int64_t aligned = BLOCK_ALIGN(s->pos);
if (!test_bit(p, aligned)) {
char tmp[BLOCK_SIZE];
stream_seek(p->original, aligned);
int r = stream_read(p->original, tmp, BLOCK_SIZE);
if (r < BLOCK_SIZE) {
if (p->size < 0) {
MP_WARN(s, "suspected EOF\n");
} else if (aligned + r < p->size) {
MP_ERR(s, "unexpected EOF\n");
return -1;
}
}
if (fseeko(p->cache_file, aligned, SEEK_SET))
return -1;
if (fwrite(tmp, r, 1, p->cache_file) != 1)
return -1;
set_bit(p, aligned, 1);
}
if (fseeko(p->cache_file, s->pos, SEEK_SET))
return -1;
// align/limit to blocks
max_len = MPMIN(max_len, BLOCK_SIZE - (s->pos % BLOCK_SIZE));
// Limit to max. known file size
max_len = MPMIN(max_len, p->size - s->pos);
return fread(buffer, 1, max_len, p->cache_file);
}
// Clamp [start, end) to range [0, size) with various fallbacks.
static void clamp_size(int size, int *start, int *end)
{
*start = MPMAX(0, *start);
*end = MPMIN(size, *end);
if (*start >= *end) {
*start = 0;
*end = 1;
}
}
static int archive_entry_seek(stream_t *s, int64_t newpos)
{
struct priv *p = s->priv;
if (p->mpa && !p->broken_seek) {
locale_t oldlocale = uselocale(p->mpa->locale);
int r = archive_seek_data(p->mpa->arch, newpos, SEEK_SET);
uselocale(oldlocale);
if (r >= 0)
return 1;
MP_WARN(s, "possibly unsupported seeking - switching to reopening\n");
p->broken_seek = true;
if (reopen_archive(s) < STREAM_OK)
return -1;
}
// libarchive can't seek in most formats.
if (newpos < s->pos) {
// Hack seeking backwards into working by reopening the archive and
// starting over.
MP_VERBOSE(s, "trying to reopen archive for performing seek\n");
if (reopen_archive(s) < STREAM_OK)
return -1;
s->pos = 0;
}
if (newpos > s->pos) {
// For seeking forwards, just keep reading data (there's no libarchive
// skip function either).
char buffer[4096];
while (newpos > s->pos) {
if (mp_cancel_test(s->cancel))
return -1;
int size = MPMIN(newpos - s->pos, sizeof(buffer));
locale_t oldlocale = uselocale(p->mpa->locale);
int r = archive_read_data(p->mpa->arch, buffer, size);
if (r <= 0) {
if (r == 0 && newpos > p->entry_size) {
MP_ERR(s, "demuxer trying to seek beyond end of archive "
"entry\n");
} else if (r == 0) {
MP_ERR(s, "end of archive entry reached while seeking\n");
} else {
MP_ERR(s, "%s\n", archive_error_string(p->mpa->arch));
}
uselocale(oldlocale);
if (mp_archive_check_fatal(p->mpa, r)) {
mp_archive_free(p->mpa);
p->mpa = NULL;
}
return -1;
}
uselocale(oldlocale);
s->pos += r;
}
}
return 1;
}
static void reselect_streams(demuxer_t *demuxer)
{
struct priv *p = demuxer->priv;
for (int n = 0; n < MPMIN(p->slave->num_streams, p->num_streams); n++) {
if (p->streams[n]) {
demuxer_select_track(p->slave, p->slave->streams[n],
demux_stream_is_selected(p->streams[n]));
}
}
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_channels_t* s = af->priv;
switch(cmd){
case AF_CONTROL_REINIT: ;
struct mp_chmap chmap;
mp_chmap_set_unknown(&chmap, s->nch);
mp_audio_set_channels(af->data, &chmap);
// Set default channel assignment
if(!s->router){
int i;
// Make sure this filter isn't redundant
if(af->data->nch == ((struct mp_audio*)arg)->nch)
return AF_DETACH;
// If mono: fake stereo
if(((struct mp_audio*)arg)->nch == 1){
s->nr = MPMIN(af->data->nch,2);
for(i=0;i<s->nr;i++){
s->route[i][FR] = 0;
s->route[i][TO] = i;
}
}
else{
s->nr = MPMIN(af->data->nch, ((struct mp_audio*)arg)->nch);
for(i=0;i<s->nr;i++){
s->route[i][FR] = i;
s->route[i][TO] = i;
}
}
}
af->data->rate = ((struct mp_audio*)arg)->rate;
mp_audio_force_interleaved_format((struct mp_audio*)arg);
mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format);
return check_routes(af,((struct mp_audio*)arg)->nch,af->data->nch);
}
return AF_UNKNOWN;
}
static void update_speed(struct af_instance *af, float speed)
{
af_scaletempo_t *s = af->priv;
s->speed = speed;
double factor = (s->speed_opt & SCALE_PITCH) ? 1.0 / s->speed : s->speed;
s->scale = factor * s->scale_nominal;
s->frames_stride_scaled = s->scale * s->frames_stride;
s->frames_stride_error = MPMIN(s->frames_stride_error, s->frames_stride_scaled);
}
static int SkipBlock(struct stream *s, const rar_block_t *hdr)
{
uint64_t size = (uint64_t)hdr->size + hdr->add_size;
while (size > 0) {
int skip = MPMIN(size, INT_MAX);
if (!stream_skip(s, skip))
return -1;
size -= skip;
}
return 0;
}
ssize_t RarRead(rar_file_t *file, void *data, size_t size)
{
size_t total = 0;
while (total < size) {
const uint64_t chunk_end = file->current_chunk->cummulated_size + file->current_chunk->size;
int max = MPMIN(MPMIN((int64_t)(size - total), (int64_t)(chunk_end - file->i_pos)), INT_MAX);
if (max <= 0)
break;
int r = stream_read(file->s, data, max);
if (r <= 0)
break;
total += r;
data = (char *)data + r;
file->i_pos += r;
if (file->i_pos >= chunk_end &&
RarSeek(file, file->i_pos))
break;
}
return total;
}
static int keys_count_matches(char *buf, int buflen) {
int count = 0;
if (buflen < 0)
buflen = strlen(buf);
for (int i = 0; i < getch2_keys.len; i++) {
keycode_st *st = &getch2_keys.map[i];
int len = MPMIN(buflen, st->len);
if (memcmp(buf, st->chars, len) == 0)
count++;
}
return count;
}
// Copy data from input frame to encode frame (because libavcodec wants a full
// AC3 frame for encoding, while filter input frames can be smaller or larger).
// Return true if the frame is complete.
static bool fill_buffer(struct af_instance *af)
{
af_ac3enc_t *s = af->priv;
af->delay = 0;
if (s->pending) {
int copy = MPMIN(s->in_samples - s->input->samples, s->pending->samples);
s->input->samples += copy;
mp_audio_copy(s->input, s->input->samples - copy, s->pending, 0, copy);
mp_audio_skip_samples(s->pending, copy);
}
update_delay(af);
return s->input->samples >= s->in_samples;
}
static bool is_compatible(int fmt1, int fmt2)
{
struct mp_imgfmt_desc d1 = mp_imgfmt_get_desc(fmt1);
struct mp_imgfmt_desc d2 = mp_imgfmt_get_desc(fmt2);
if (d1.num_planes < d2.num_planes)
return false;
if (!(d1.flags & MP_IMGFLAG_BYTE_ALIGNED) ||
!(d2.flags & MP_IMGFLAG_BYTE_ALIGNED))
return false;
for (int n = 0; n < MPMIN(d1.num_planes, d2.num_planes); n++) {
if (d1.bytes[n] != d2.bytes[n])
return false;
if (d1.xs[n] != d2.xs[n] || d1.ys[n] != d2.ys[n])
return false;
}
return true;
}
// Copy data from input frame to encode frame (because libavcodec wants a full
// AC3 frame for encoding, while filter input frames can be smaller or larger).
// Return true if the frame is complete.
static bool fill_buffer(struct af_instance *af)
{
af_ac3enc_t *s = af->priv;
af->delay = 0;
if (s->pending) {
if (!mp_audio_is_writeable(s->input))
assert(s->input->samples == 0); // we can't have sent a partial frame
mp_audio_realloc_min(s->input, s->in_samples);
int copy = MPMIN(s->in_samples - s->input->samples, s->pending->samples);
s->input->samples += copy;
mp_audio_copy(s->input, s->input->samples - copy, s->pending, 0, copy);
mp_audio_skip_samples(s->pending, copy);
}
update_delay(af);
return s->input->samples >= s->in_samples;
}
void mp_set_avcodec_threads(struct mp_log *l, AVCodecContext *avctx, int threads)
{
if (threads == 0) {
threads = av_cpu_count();
if (threads < 1) {
mp_warn(l, "Could not determine thread count to use, defaulting to 1.\n");
threads = 1;
} else {
mp_verbose(l, "Detected %d logical cores.\n", threads);
if (threads > 1)
threads += 1; // extra thread for better load balancing
}
// Apparently some libavcodec versions have or had trouble with more
// than 16 threads, and/or print a warning when using > 16.
threads = MPMIN(threads, 16);
}
mp_verbose(l, "Requesting %d threads for decoding.\n", threads);
avctx->thread_count = threads;
}
static float compute_replaygain(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
float rgain = 1.0;
struct replaygain_data *rg = NULL;
struct track *track = mpctx->current_track[0][STREAM_AUDIO];
if (track)
rg = track->stream->codec->replaygain_data;
if (opts->rgain_mode && rg) {
MP_VERBOSE(mpctx, "Replaygain: Track=%f/%f Album=%f/%f\n",
rg->track_gain, rg->track_peak,
rg->album_gain, rg->album_peak);
float gain, peak;
if (opts->rgain_mode == 1) {
gain = rg->track_gain;
peak = rg->track_peak;
} else {
gain = rg->album_gain;
peak = rg->album_peak;
}
gain += opts->rgain_preamp;
rgain = db_gain(gain);
MP_VERBOSE(mpctx, "Applying replay-gain: %f\n", rgain);
if (!opts->rgain_clip) { // clipping prevention
rgain = MPMIN(rgain, 1.0 / peak);
MP_VERBOSE(mpctx, "...with clipping prevention: %f\n", rgain);
}
} else if (opts->rgain_fallback) {
rgain = db_gain(opts->rgain_fallback);
MP_VERBOSE(mpctx, "Applying fallback gain: %f\n", rgain);
}
return rgain;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *priv = ao->priv;
int accepted;
resume(ao);
if (priv->buffered <= 0)
priv->buffered = priv->latency; // emulate fixed latency
priv->playing_final = flags & AOPLAY_FINAL_CHUNK;
if (priv->playing_final) {
// Last audio chunk - don't round to outburst.
accepted = MPMIN(priv->buffersize - priv->buffered, samples);
} else {
int maxbursts = (priv->buffersize - priv->buffered) / priv->outburst;
int playbursts = samples / priv->outburst;
int bursts = playbursts > maxbursts ? maxbursts : playbursts;
accepted = bursts * priv->outburst;
}
priv->buffered += accepted;
return accepted;
}
struct timespec mp_time_us_to_timespec(int64_t time_us)
{
struct timespec ts;
get_realtime(&ts);
// We don't know what time source mp_time_us() uses, but usually it's not
// CLOCK_REALTIME - so we have to remap the times.
int64_t unow = mp_time_us();
int64_t diff_us = time_us - unow;
int64_t diff_secs = diff_us / (1000L * 1000L);
long diff_nsecs = (diff_us - diff_secs * (1000L * 1000L)) * 1000L;
if (diff_nsecs < 0) {
diff_secs -= 1;
diff_nsecs += 1000000000L;
}
if (diff_nsecs + ts.tv_nsec >= 1000000000UL) {
diff_secs += 1;
diff_nsecs -= 1000000000UL;
}
// OSX can't deal with large timeouts. Also handles tv_sec/time_t overflows.
diff_secs = MPMIN(diff_secs, 10000000);
ts.tv_sec += diff_secs;
ts.tv_nsec += diff_nsecs;
return ts;
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* audio, int flags)
{
struct mp_audio *out = af->data;
af_ac3enc_t *s = af->priv;
int num_frames = (audio->samples + mp_audio_buffer_samples(s->pending))
/ s->in_samples;
int max_out_samples = s->out_samples * num_frames;
mp_audio_realloc_min(out, max_out_samples);
out->samples = 0;
while (audio->samples > 0) {
int ret;
int consumed_pending = 0;
struct mp_audio in_frame;
int pending = mp_audio_buffer_samples(s->pending);
if (pending == 0 && audio->samples >= s->in_samples) {
in_frame = *audio;
mp_audio_skip_samples(audio, s->in_samples);
} else {
if (pending > 0 && pending < s->in_samples) {
struct mp_audio tmp = *audio;
tmp.samples = MPMIN(tmp.samples, s->in_samples);
mp_audio_buffer_append(s->pending, &tmp);
mp_audio_skip_samples(audio, tmp.samples);
}
mp_audio_buffer_peek(s->pending, &in_frame);
if (in_frame.samples < s->in_samples)
break;
consumed_pending = s->in_samples;
}
in_frame.samples = s->in_samples;
AVFrame *frame = av_frame_alloc();
if (!frame) {
MP_FATAL(af, "Could not allocate memory \n");
return -1;
}
frame->nb_samples = s->in_samples;
frame->format = s->lavc_actx->sample_fmt;
frame->channel_layout = s->lavc_actx->channel_layout;
assert(in_frame.num_planes <= AV_NUM_DATA_POINTERS);
frame->extended_data = frame->data;
for (int n = 0; n < in_frame.num_planes; n++)
frame->data[n] = in_frame.planes[n];
frame->linesize[0] = s->in_samples * audio->sstride;
int ok;
ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
av_frame_free(&frame);
if (ret < 0 || !ok) {
MP_FATAL(af, "Encode failed.\n");
return -1;
}
mp_audio_buffer_skip(s->pending, consumed_pending);
MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
s->pkt.size, mp_audio_buffer_samples(s->pending));
int frame_size = s->pkt.size;
int header_len = 0;
char hdr[8];
if (s->cfg_add_iec61937_header && s->pkt.size > 5) {
int bsmod = s->pkt.data[5] & 0x7;
int len = frame_size;
frame_size = AC3_FRAME_SIZE * 2 * 2;
header_len = 8;
AV_WB16(hdr, 0xF872); // iec 61937 syncword 1
AV_WB16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
hdr[4] = bsmod; // bsmod
hdr[5] = 0x01; // data-type ac3
AV_WB16(hdr + 6, len << 3); // number of bits in payload
}
size_t max_size = (max_out_samples - out->samples) * out->sstride;
if (frame_size > max_size)
abort();
char *buf = (char *)out->planes[0] + out->samples * out->sstride;
memcpy(buf, hdr, header_len);
memcpy(buf + header_len, s->pkt.data, s->pkt.size);
memset(buf + header_len + s->pkt.size, 0,
frame_size - (header_len + s->pkt.size));
out->samples += frame_size / out->sstride;
}
mp_audio_buffer_append(s->pending, audio);
*audio = *out;
return 0;
}
// Filter len bytes of input, put result into outbuf.
static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
int len)
{
bool format_change = false;
int error = 0;
assert(len > 0); // would break EOF logic below
while (mp_audio_buffer_samples(da->decode_buffer) < len) {
// Check for a format change
struct mp_audio config;
mp_audio_buffer_get_format(da->decode_buffer, &config);
format_change = !mp_audio_config_equals(&da->decoded, &config);
if (format_change) {
error = AD_EOF; // drain remaining data left in the current buffer
break;
}
if (da->decoded.samples > 0) {
int copy = MPMIN(da->decoded.samples, len);
struct mp_audio append = da->decoded;
append.samples = copy;
mp_audio_buffer_append(da->decode_buffer, &append);
mp_audio_skip_samples(&da->decoded, copy);
da->pts_offset += copy;
continue;
}
error = da->ad_driver->decode_packet(da);
if (error < 0)
break;
}
if (error == AD_WAIT)
return error;
// Filter
struct mp_audio filter_data;
mp_audio_buffer_peek(da->decode_buffer, &filter_data);
filter_data.rate = da->afilter->input.rate; // due to playback speed change
len = MPMIN(filter_data.samples, len);
filter_data.samples = len;
bool eof = error == AD_EOF && filter_data.samples == 0;
if (af_filter(da->afilter, &filter_data, eof ? AF_FILTER_FLAG_EOF : 0) < 0)
return AD_ERR;
mp_audio_buffer_append(outbuf, &filter_data);
if (error == AD_EOF && filter_data.samples > 0)
error = 0; // don't end playback yet
// remove processed data from decoder buffer:
mp_audio_buffer_skip(da->decode_buffer, len);
// if format was changed, and all data was drained, execute the format change
if (format_change && eof) {
error = AD_NEW_FMT;
if (!reinit_audio_buffer(da))
error = AD_ERR; // switch to invalid format
}
return error;
}
static int init(struct ao *ao)
{
if (SDL_WasInit(SDL_INIT_AUDIO)) {
MP_ERR(ao, "already initialized\n");
return -1;
}
struct priv *priv = ao->priv;
if (SDL_InitSubSystem(SDL_INIT_AUDIO)) {
if (!ao->probing)
MP_ERR(ao, "SDL_Init failed\n");
uninit(ao);
return -1;
}
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_waveext_def(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) {
uninit(ao);
return -1;
}
ao->format = af_fmt_from_planar(ao->format);
SDL_AudioSpec desired, obtained;
desired.format = AUDIO_S16SYS;
for (int n = 0; fmtmap[n][0]; n++) {
if (ao->format == fmtmap[n][0]) {
desired.format = fmtmap[n][1];
break;
}
}
desired.freq = ao->samplerate;
desired.channels = ao->channels.num;
desired.samples = MPMIN(32768, ceil_power_of_two(ao->samplerate *
priv->buflen));
desired.callback = audio_callback;
desired.userdata = ao;
MP_VERBOSE(ao, "requested format: %d Hz, %d channels, %x, "
"buffer size: %d samples\n",
(int) desired.freq, (int) desired.channels,
(int) desired.format, (int) desired.samples);
obtained = desired;
if (SDL_OpenAudio(&desired, &obtained)) {
if (!ao->probing)
MP_ERR(ao, "could not open audio: %s\n", SDL_GetError());
uninit(ao);
return -1;
}
MP_VERBOSE(ao, "obtained format: %d Hz, %d channels, %x, "
"buffer size: %d samples\n",
(int) obtained.freq, (int) obtained.channels,
(int) obtained.format, (int) obtained.samples);
// The sample count is usually the number of samples the callback requests,
// which we assume is the period size. Normally, ao.c will allocate a large
// enough buffer. But in case the period size should be pathologically
// large, this will help.
ao->device_buffer = 3 * obtained.samples;
ao->format = 0;
for (int n = 0; fmtmap[n][0]; n++) {
if (obtained.format == fmtmap[n][1]) {
ao->format = fmtmap[n][0];
break;
}
}
if (!ao->format) {
if (!ao->probing)
MP_ERR(ao, "could not find matching format\n");
uninit(ao);
return -1;
}
if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, obtained.channels)) {
uninit(ao);
return -1;
}
ao->samplerate = obtained.freq;
priv->paused = 1;
return 1;
}
static int audio_start_sync(struct MPContext *mpctx, int playsize)
{
struct ao *ao = mpctx->ao;
struct MPOpts *opts = mpctx->opts;
sh_audio_t * const sh_audio = mpctx->sh_audio;
int res;
// Timing info may not be set without
res = decode_audio(sh_audio, &ao->buffer, 1);
if (res < 0)
return res;
int bytes;
bool did_retry = false;
double written_pts;
double bps = ao->bps / opts->playback_speed;
bool hrseek = mpctx->hrseek_active; // audio only hrseek
mpctx->hrseek_active = false;
while (1) {
written_pts = written_audio_pts(mpctx);
double ptsdiff;
if (hrseek)
ptsdiff = written_pts - mpctx->hrseek_pts;
else
ptsdiff = written_pts - mpctx->sh_video->pts - mpctx->delay
- mpctx->audio_delay;
bytes = ptsdiff * bps;
bytes -= bytes % (ao->channels.num * af_fmt2bits(ao->format) / 8);
// ogg demuxers give packets without timing
if (written_pts <= 1 && sh_audio->pts == MP_NOPTS_VALUE) {
if (!did_retry) {
// Try to read more data to see packets that have pts
res = decode_audio(sh_audio, &ao->buffer, ao->bps);
if (res < 0)
return res;
did_retry = true;
continue;
}
bytes = 0;
}
if (fabs(ptsdiff) > 300 || isnan(ptsdiff)) // pts reset or just broken?
bytes = 0;
if (bytes > 0)
break;
mpctx->syncing_audio = false;
int a = MPMIN(-bytes, MPMAX(playsize, 20000));
res = decode_audio(sh_audio, &ao->buffer, a);
bytes += ao->buffer.len;
if (bytes >= 0) {
memmove(ao->buffer.start,
ao->buffer.start + ao->buffer.len - bytes, bytes);
ao->buffer.len = bytes;
if (res < 0)
return res;
return decode_audio(sh_audio, &ao->buffer, playsize);
}
ao->buffer.len = 0;
if (res < 0)
return res;
}
if (hrseek)
// Don't add silence in audio-only case even if position is too late
return 0;
int fillbyte = 0;
if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_US)
fillbyte = 0x80;
if (bytes >= playsize) {
/* This case could fall back to the one below with
* bytes = playsize, but then silence would keep accumulating
* in a_out_buffer if the AO accepts less data than it asks for
* in playsize. */
char *p = malloc(playsize);
memset(p, fillbyte, playsize);
write_to_ao(mpctx, p, playsize, 0, written_pts - bytes / bps);
free(p);
return ASYNC_PLAY_DONE;
}
mpctx->syncing_audio = false;
decode_audio_prepend_bytes(&ao->buffer, bytes, fillbyte);
return decode_audio(sh_audio, &ao->buffer, playsize);
}
