void audio_callback(void *userdata, Uint8 *stream, int len)
{
AudioState *audio = userdata;
int len1, audio_size;
double pts;
while(len > 0) {
if(audio->audio_buf_index >= audio->audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = decode_audio(audio, audio->audioBuf, sizeof(audio->audioBuf), &pts);
if(audio_size < 0) {
/* If error, output silence */
audio->audio_buf_size = 1024;
memset(audio->audioBuf, 0, audio->audio_buf_size);
} else {
audio->audio_buf_size = audio_size;
}
audio->audio_buf_index = 0;
}
len1 = audio->audio_buf_size - audio->audio_buf_index;
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)audio->audioBuf + audio->audio_buf_index, len1);
len -= len1;
stream += len1;
audio->audio_buf_index += len1;
}
}
// it is running in sound mixer thread
void as_netstream::audio_callback(Uint8* stream, int len)
{
while (len > 0 && m_status == PLAY)
{
// decode sound
if (m_sound == NULL)
{
gc_ptr<av_packet> audio;
if (m_aq.pop(&audio))
{
Sint16* sample;
int size;
decode_audio(audio->get_packet(), &sample, &size);
m_sound = new decoded_sound(sample, size);
continue;
}
break;
}
else
{
int n = m_sound->extract(stream, len);
stream += n;
len -= n;
if (m_sound->size() == 0)
{
m_sound = NULL;
}
}
}
}
SCM ffmpeg_decode_audio_video(SCM scm_self)
{
SCM retval = SCM_BOOL_F;
struct ffmpeg_t *self = get_self(scm_self);
if (!is_input_context(self))
scm_misc_error("ffmpeg-decode-audio/video", "Attempt to read frame from FFmpeg output video", SCM_EOL);
while (scm_is_false(retval)) {
if (packet_empty(self)) read_packet(self);
int reading_cache = packet_empty(self);
if (self->pkt.stream_index == self->audio_stream_idx) {
av_frame_unref(self->audio_target_frame);
retval = decode_audio(self, &self->pkt, self->audio_target_frame);
} else if (self->pkt.stream_index == self->video_stream_idx) {
av_frame_unref(self->video_target_frame);
retval = decode_video(self, &self->pkt, self->video_target_frame);
} else
consume_packet_data(&self->pkt, self->pkt.size);
if (scm_is_false(retval) && reading_cache) break;
};
return retval;
}
static int init(sh_audio_t *sh_audio)
{
int ret=decode_audio(sh_audio,sh_audio->a_buffer,4096,sh_audio->a_buffer_size);
if(ret<0){
mp_msg(MSGT_DECAUDIO,MSGL_INFO,"ACM decoding error: %d\n",ret);
return 0;
}
sh_audio->a_buffer_len=ret;
return 1;
}
void *decoder_thread(void *arg)
{
PlayerCtx *ctx = arg;
AVPacket packet;
AVPacket cpacket;
size_t decoded_bytes;
int seekid = ctx->cur_seekid;
printf("Decoder thread started\n");
memset(&packet, 0, sizeof(packet));
memset(&cpacket, 0, sizeof(cpacket));
while (!ctx->exit) {
int new_packet = 0;
if (cpacket.size == 0) {
if (packet.data)
av_free_packet(&packet);
pthread_mutex_lock(&ctx->seek_mutex);
if (ctx->cur_seekid > seekid) {
printf("Seek! %f\n", ctx->seek_pos);
av_seek_frame(ctx->fmt_ctx, -1, (int64_t)(ctx->seek_pos * AV_TIME_BASE), 0);
seekid = ctx->cur_seekid;
// HACK! Avoid deadlock by waking up the video waiter
pthread_mutex_lock(&ctx->v_buf_mutex);
pthread_cond_signal(&ctx->v_buf_not_empty);
pthread_mutex_unlock(&ctx->v_buf_mutex);
if (ctx->audio_idx != -1)
avcodec_flush_buffers(ctx->a_codec_ctx);
avcodec_flush_buffers(ctx->v_codec_ctx);
}
if (av_read_frame(ctx->fmt_ctx, &packet) < 0) {
fprintf(stderr, "EOF!\n");
push_eof(ctx, seekid);
pthread_cond_wait(&ctx->seek_cond, &ctx->seek_mutex);
pthread_mutex_unlock(&ctx->seek_mutex);
continue;
}
pthread_mutex_unlock(&ctx->seek_mutex);
cpacket = packet;
new_packet = 1;
}
if (ctx->audio_idx != -1 && cpacket.stream_index == ctx->audio_idx) {
decoded_bytes = decode_audio(ctx, &cpacket, new_packet, seekid);
} else if (cpacket.stream_index == ctx->video_idx) {
decoded_bytes = decode_video(ctx, &cpacket, new_packet, seekid);
} else {
decoded_bytes = cpacket.size;
}
cpacket.data += decoded_bytes;
cpacket.size -= decoded_bytes;
}
return NULL;
}
static int control(sh_audio_t *sh, int cmd, void* arg, ...) {
if (cmd == ADCTRL_RESYNC_STREAM) {
unsigned char *buf = malloc(MAX_FRAMESIZE);
int i;
int nr_ok = 0;
for (i = 0; i < MAX_SEEK_DISCARD; i++) {
int len = decode_audio(sh, buf, 0, MAX_FRAMESIZE);
if (check_clip(buf, len)) nr_ok++; else nr_ok = 0;
if (nr_ok > MIN_SEEK_GOOD) break;
}
free(buf);
}
return CONTROL_UNKNOWN;
}
static int init(sh_audio_t *sh_audio)
{
struct MPOpts *opts = sh_audio->opts;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
if (sh_audio->codec->dll) {
lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
if (!lavc_codec) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
"Cannot find codec '%s' in libavcodec...\n",
sh_audio->codec->dll);
return 0;
}
} else if (!sh_audio->libav_codec_id) {
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "No Libav codec ID known. "
"Generic lavc decoder is not applicable.\n");
return 0;
} else {
lavc_codec = avcodec_find_decoder(sh_audio->libav_codec_id);
if (!lavc_codec) {
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Libavcodec has no decoder "
"for this codec\n");
return 0;
}
}
sh_audio->codecname = lavc_codec->long_name;
if (!sh_audio->codecname)
sh_audio->codecname = lavc_codec->name;
struct priv *ctx = talloc_zero(NULL, struct priv);
sh_audio->context = ctx;
lavc_context = avcodec_alloc_context3(lavc_codec);
ctx->avctx = lavc_context;
ctx->avframe = avcodec_alloc_frame();
// Always try to set - option only exists for AC3 at the moment
av_opt_set_double(lavc_context, "drc_scale", opts->drc_level,
AV_OPT_SEARCH_CHILDREN);
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
if (sh_audio->wf) {
lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->request_channels = opts->audio_output_channels;
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
/* alloc extra data */
if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->wf->cbSize;
memcpy(lavc_context->extradata, sh_audio->wf + 1,
lavc_context->extradata_size);
}
// for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata &&
!lavc_context->extradata) {
lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
/* open it */
if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
uninit(sh_audio);
return 0;
}
mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
lavc_codec->name);
if (sh_audio->format == 0x3343414D) {
// MACE 3:1
sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
} else if (sh_audio->format == 0x3643414D) {
// MACE 6:1
sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet
sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
}
// 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;
break;
}
if (++tries >= 5) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"ad_ffmpeg: initial decode failed\n");
uninit(sh_audio);
return 0;
}
}
sh_audio->i_bps = lavc_context->bit_rate / 8;
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
return 1;
}
int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
struct mgcp_rtp_end *dst_end,
char *data, int *len, int buf_size)
{
struct mgcp_process_rtp_state *state;
const size_t rtp_hdr_size = sizeof(struct rtp_hdr);
struct rtp_hdr *rtp_hdr = (struct rtp_hdr *) data;
char *payload_data = (char *) &rtp_hdr->data[0];
int payload_len = *len - rtp_hdr_size;
uint8_t *src = (uint8_t *)payload_data;
uint8_t *dst = (uint8_t *)payload_data;
size_t nbytes = payload_len;
size_t nsamples;
size_t max_samples;
uint32_t ts_no;
int rc;
state = check_transcode_state(endp, dst_end, rtp_hdr);
if (!state)
return 0;
if (state->src_fmt == state->dst_fmt) {
if (!state->dst_packet_duration)
return 0;
/* TODO: repackage without transcoding */
}
/* If the remaining samples do not fit into a fixed ptime,
* a) discard them, if the next packet is much later
* b) add silence and * send it, if the current packet is not
* yet too late
* c) append the sample data, if the timestamp matches exactly
*/
/* TODO: check payload type (-> G.711 comfort noise) */
if (payload_len > 0) {
ts_no = ntohl(rtp_hdr->timestamp);
if (!state->is_running) {
state->next_seq = ntohs(rtp_hdr->sequence);
state->next_time = ts_no;
state->is_running = 1;
}
if (state->sample_cnt > 0) {
int32_t delta = ts_no - state->next_time;
/* TODO: check sequence? reordering? packet loss? */
if (delta > state->sample_cnt) {
/* There is a time gap between the last packet
* and the current one. Just discard the
* partial data that is left in the buffer.
* TODO: This can be improved by adding silence
* instead if the delta is small enough.
*/
LOGP(DMGCP, LOGL_NOTICE,
"0x%x dropping sample buffer due delta=%d sample_cnt=%d\n",
ENDPOINT_NUMBER(endp), delta, state->sample_cnt);
state->sample_cnt = 0;
state->next_time = ts_no;
} else if (delta < 0) {
LOGP(DMGCP, LOGL_NOTICE,
"RTP time jumps backwards, delta = %d, "
"discarding buffered samples\n",
delta);
state->sample_cnt = 0;
state->sample_offs = 0;
return -EAGAIN;
}
/* Make sure the samples start without offset */
if (state->sample_offs && state->sample_cnt)
memmove(&state->samples[0],
&state->samples[state->sample_offs],
state->sample_cnt *
sizeof(state->samples[0]));
}
state->sample_offs = 0;
/* Append decoded audio to samples */
decode_audio(state, &src, &nbytes);
if (nbytes > 0)
LOGP(DMGCP, LOGL_NOTICE,
"Skipped audio frame in RTP packet: %d octets\n",
nbytes);
} else
ts_no = state->next_time;
if (state->sample_cnt < state->dst_packet_duration)
return -EAGAIN;
max_samples =
state->dst_packet_duration ?
state->dst_packet_duration : state->sample_cnt;
nsamples = state->sample_cnt;
rc = encode_audio(state, dst, buf_size, max_samples);
/*
* There were no samples to encode?
* TODO: how does this work for comfort noise?
*/
if (rc == 0)
return -ENOMSG;
/* Any other error during the encoding */
if (rc < 0)
return rc;
nsamples -= state->sample_cnt;
*len = rtp_hdr_size + rc;
rtp_hdr->sequence = htons(state->next_seq);
rtp_hdr->timestamp = htonl(ts_no);
state->next_seq += 1;
state->next_time = ts_no + nsamples;
/*
* XXX: At this point we should always have consumed
* samples. So doing OSMO_ASSERT(nsamples > 0) and returning
* rtp_hdr_size should be fine.
*/
return nsamples ? rtp_hdr_size : 0;
}
int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
{
struct MPOpts *opts = mpctx->opts;
struct ao *ao = mpctx->ao;
int playsize;
int playflags = 0;
bool audio_eof = false;
bool signal_eof = false;
bool partial_fill = false;
sh_audio_t * const sh_audio = mpctx->sh_audio;
bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK);
int unitsize = ao->channels.num * af_fmt2bits(ao->format) / 8;
if (mpctx->paused)
playsize = 1; // just initialize things (audio pts at least)
else
playsize = ao_get_space(ao);
// Coming here with hrseek_active still set means audio-only
if (!mpctx->sh_video || !mpctx->sync_audio_to_video)
mpctx->syncing_audio = false;
if (!opts->initial_audio_sync || !modifiable_audio_format) {
mpctx->syncing_audio = false;
mpctx->hrseek_active = false;
}
int res;
if (mpctx->syncing_audio || mpctx->hrseek_active)
res = audio_start_sync(mpctx, playsize);
else
res = decode_audio(sh_audio, &ao->buffer, playsize);
if (res < 0) { // EOF, error or format change
if (res == -2) {
/* The format change isn't handled too gracefully. A more precise
* implementation would require draining buffered old-format audio
* while displaying video, then doing the output format switch.
*/
if (!mpctx->opts->gapless_audio)
uninit_player(mpctx, INITIALIZED_AO);
reinit_audio_chain(mpctx);
return -1;
} else if (res == ASYNC_PLAY_DONE)
return 0;
else if (demux_stream_eof(mpctx->sh_audio->gsh))
audio_eof = true;
}
if (endpts != MP_NOPTS_VALUE && modifiable_audio_format) {
double bytes = (endpts - written_audio_pts(mpctx) + mpctx->audio_delay)
* ao->bps / opts->playback_speed;
if (playsize > bytes) {
playsize = MPMAX(bytes, 0);
audio_eof = true;
partial_fill = true;
}
}
assert(ao->buffer.len % unitsize == 0);
if (playsize > ao->buffer.len) {
partial_fill = true;
playsize = ao->buffer.len;
}
playsize -= playsize % unitsize;
if (!playsize)
return partial_fill && audio_eof ? -2 : -partial_fill;
if (audio_eof && partial_fill) {
if (opts->gapless_audio) {
// With gapless audio, delay this to ao_uninit. There must be only
// 1 final chunk, and that is handled when calling ao_uninit().
signal_eof = true;
} else {
playflags |= AOPLAY_FINAL_CHUNK;
}
}
assert(ao->buffer_playable_size <= ao->buffer.len);
int played = write_to_ao(mpctx, ao->buffer.start, playsize, playflags,
written_audio_pts(mpctx));
ao->buffer_playable_size = playsize - played;
if (played > 0) {
ao->buffer.len -= played;
memmove(ao->buffer.start, ao->buffer.start + played, ao->buffer.len);
} else if (!mpctx->paused && audio_eof && ao_get_delay(ao) < .04) {
// Sanity check to avoid hanging in case current ao doesn't output
// partial chunks and doesn't check for AOPLAY_FINAL_CHUNK
signal_eof = true;
}
return signal_eof ? -2 : -partial_fill;
}
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);
}
static void *toxav_decoding(void *arg)
{
void **pp = arg;
ToxAv *av = pp[0];
CallSpecific *call = pp[1];
free(pp);
while (1) {
DECODE_PACKET *p;
_Bool video = 0;
pthread_mutex_lock(&call->decode_cond_mutex);
if (call->exit) {
break;
}
uint8_t r;
/* first check for available packets, otherwise wait for condition*/
r = call->audio_decode_read;
p = call->audio_decode_queue[r];
if (!p) {
r = call->video_decode_read;
p = call->video_decode_queue[r];
if (!p) {
pthread_cond_wait(&call->decode_cond, &call->decode_cond_mutex);
r = call->audio_decode_read;
p = call->audio_decode_queue[r];
if (!p) {
r = call->video_decode_read;
p = call->video_decode_queue[r];
video = 1;
}
} else {
video = 1;
}
}
if (video) {
if (p) {
call->video_decode_queue[r] = NULL;
call->video_decode_read = (r + 1) % VIDEO_DECODE_QUEUE_SIZE;
}
} else {
call->audio_decode_queue[r] = NULL;
call->audio_decode_read = (r + 1) % AUDIO_DECODE_QUEUE_SIZE;
}
pthread_mutex_unlock(&call->decode_cond_mutex);
if (p) {
if (video) {
decode_video(av, call, p);
} else {
decode_audio(av, call, p);
}
}
}
call->exit = 0;
pthread_cond_signal(&call->decode_cond);
pthread_mutex_unlock(&call->decode_cond_mutex);
return NULL;
}
static int init(sh_audio_t *sh_audio)
{
struct MPOpts *opts = sh_audio->opts;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
return 0;
}
lavc_context = avcodec_alloc_context();
sh_audio->context=lavc_context;
lavc_context->drc_scale = opts->drc_level;
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
if(sh_audio->wf){
lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->request_channels = opts->audio_output_channels;
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
/* alloc extra data */
if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->wf->cbSize;
memcpy(lavc_context->extradata, sh_audio->wf + 1,
lavc_context->extradata_size);
}
// for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
{
lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
/* open it */
if (avcodec_open(lavc_context, lavc_codec) < 0) {
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
return 0;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
// printf("\nFOURCC: 0x%X\n",sh_audio->format);
if(sh_audio->format==0x3343414D){
// MACE 3:1
sh_audio->ds->ss_div = 2*3; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
} else
if(sh_audio->format==0x3643414D){
// MACE 6:1
sh_audio->ds->ss_div = 2*6; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
}
// 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;
break;
}
if (++tries >= 5) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"ad_ffmpeg: initial decode failed\n");
return 0;
}
}
sh_audio->i_bps=lavc_context->bit_rate/8;
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
switch (lavc_context->sample_fmt) {
case SAMPLE_FMT_U8:
case SAMPLE_FMT_S16:
case SAMPLE_FMT_S32:
case SAMPLE_FMT_FLT:
break;
default:
return 0;
}
return 1;
}
static int init(sh_audio_t *sh_audio)
{
int x;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
if(!avcodec_inited){
avcodec_init();
avcodec_register_all();
avcodec_inited=1;
}
lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
return 0;
}
lavc_context = avcodec_alloc_context();
sh_audio->context=lavc_context;
if(sh_audio->wf){
lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
/* alloc extra data */
if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->wf->cbSize;
memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX),
lavc_context->extradata_size);
}
// for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
{
lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
/* open it */
if (avcodec_open(lavc_context, lavc_codec) < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
return 0;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n");
// printf("\nFOURCC: 0x%X\n",sh_audio->format);
if(sh_audio->format==0x3343414D){
// MACE 3:1
sh_audio->ds->ss_div = 2*3; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
} else
if(sh_audio->format==0x3643414D){
// MACE 6:1
sh_audio->ds->ss_div = 2*6; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
}
// Decode at least 1 byte: (to get header filled)
x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
if(x>0) sh_audio->a_buffer_len=x;
sh_audio->channels=lavc_context->channels;
sh_audio->samplerate=lavc_context->sample_rate;
sh_audio->i_bps=lavc_context->bit_rate/8;
if(sh_audio->wf){
// If the decoder uses the wrong number of channels all is lost anyway.
// sh_audio->channels=sh_audio->wf->nChannels;
if (sh_audio->wf->nSamplesPerSec)
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
if (sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
}
sh_audio->samplesize=2;
return 1;
}
static int init(sh_audio_t *sh_audio)
{
int tries = 0;
int x;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
if(!avcodec_initialized){
avcodec_init();
avcodec_register_all();
avcodec_initialized=1;
}
lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
return 0;
}
lavc_context = avcodec_alloc_context();
sh_audio->context=lavc_context;
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
if(sh_audio->wf){
lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->request_channels = audio_output_channels;
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_type = CODEC_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
/* alloc extra data */
if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->wf->cbSize;
memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX),
lavc_context->extradata_size);
}
// for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
{
lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
/* open it */
if (avcodec_open(lavc_context, lavc_codec) < 0) {
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
return 0;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
// printf("\nFOURCC: 0x%X\n",sh_audio->format);
if(sh_audio->format==0x3343414D){
// MACE 3:1
sh_audio->ds->ss_div = 2*3; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
} else
if(sh_audio->format==0x3643414D){
// MACE 6:1
sh_audio->ds->ss_div = 2*6; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
}
// Decode at least 1 byte: (to get header filled)
do {
x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
} while (x <= 0 && tries++ < 5);
if(x>0) sh_audio->a_buffer_len=x;
sh_audio->channels=lavc_context->channels;
sh_audio->samplerate=lavc_context->sample_rate;
sh_audio->i_bps=lavc_context->bit_rate/8;
switch (lavc_context->sample_fmt) {
case SAMPLE_FMT_U8: sh_audio->sample_format = AF_FORMAT_U8; break;
case SAMPLE_FMT_S16: sh_audio->sample_format = AF_FORMAT_S16_NE; break;
case SAMPLE_FMT_S32: sh_audio->sample_format = AF_FORMAT_S32_NE; break;
case SAMPLE_FMT_FLT: sh_audio->sample_format = AF_FORMAT_FLOAT_NE; break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
return 0;
}
/* If the audio is AAC the container level data may be unreliable
* because of SBR handling problems (possibly half real sample rate at
* container level). Default AAC decoding with ad_faad has used codec-level
* values for a long time without generating complaints so it should be OK.
*/
if (sh_audio->wf && lavc_context->codec_id != CODEC_ID_AAC) {
// If the decoder uses the wrong number of channels all is lost anyway.
// sh_audio->channels=sh_audio->wf->nChannels;
if (sh_audio->wf->nSamplesPerSec)
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
if (sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
}
sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
return 1;
}
static gboolean
process (GeglOperation *operation,
GeglBuffer *output,
const GeglRectangle *result,
gint level)
{
GeglProperties *o = GEGL_PROPERTIES (operation);
Priv *p = (Priv*)o->user_data;
{
if (p->video_fcontext && !decode_frame (operation, o->frame))
{
long sample_start = 0;
if (p->audio_stream)
{
int sample_count;
gegl_audio_fragment_set_sample_rate (o->audio, p->audio_stream->codec->sample_rate);
gegl_audio_fragment_set_channels (o->audio, 2);
gegl_audio_fragment_set_channel_layout (o->audio, GEGL_CH_LAYOUT_STEREO);
sample_count = samples_per_frame (o->frame,
o->frame_rate, p->audio_stream->codec->sample_rate,
&sample_start);
gegl_audio_fragment_set_sample_count (o->audio, sample_count);
decode_audio (operation, p->prevpts, p->prevpts + 5.0);
{
int i;
for (i = 0; i < sample_count; i++)
{
get_sample_data (p, sample_start + i, &o->audio->data[0][i],
&o->audio->data[1][i]);
}
}
}
if (p->video_stream->codec->pix_fmt == AV_PIX_FMT_RGB24)
{
GeglRectangle extent = {0,0,p->width,p->height};
gegl_buffer_set (output, &extent, 0, babl_format("R'G'B' u8"), p->lavc_frame->data[0], GEGL_AUTO_ROWSTRIDE);
}
else
{
struct SwsContext *img_convert_ctx;
GeglRectangle extent = {0,0,p->width,p->height};
img_convert_ctx = sws_getContext(p->width, p->height, p->video_stream->codec->pix_fmt,
p->width, p->height, AV_PIX_FMT_RGB24,
SWS_BICUBIC, NULL, NULL, NULL);
if (!p->rgb_frame)
p->rgb_frame = alloc_picture (AV_PIX_FMT_RGB24, p->width, p->height);
sws_scale (img_convert_ctx, (void*)p->lavc_frame->data,
p->lavc_frame->linesize, 0, p->height, p->rgb_frame->data, p->rgb_frame->linesize);
gegl_buffer_set (output, &extent, 0, babl_format("R'G'B' u8"), p->rgb_frame->data[0], GEGL_AUTO_ROWSTRIDE);
sws_freeContext (img_convert_ctx);
}
}
}
return TRUE;
}
size_t FFmpegDecoderAudio::decodeFrame(void * const buffer, const size_t size)
{
for (;;)
{
// Decode current packet
while (m_bytes_remaining > 0)
{
int data_size = size;
const int bytes_decoded = decode_audio(m_context, reinterpret_cast<int16_t*>(buffer), &data_size, m_packet_data, m_bytes_remaining);
if (bytes_decoded < 0)
{
// if error, skip frame
m_bytes_remaining = 0;
break;
}
m_bytes_remaining -= bytes_decoded;
m_packet_data += bytes_decoded;
// If we have some data, return it and come back for more later.
if (data_size > 0)
return data_size;
}
// Get next packet
if (m_packet.valid())
m_packet.clear();
if (m_exit)
return 0;
bool is_empty = true;
m_packet = m_packets.tryPop(is_empty);
if (is_empty)
return 0;
if (m_packet.type == FFmpegPacket::PACKET_DATA)
{
if (m_packet.packet.pts != int64_t(AV_NOPTS_VALUE))
{
const double pts = av_q2d(m_stream->time_base) * m_packet.packet.pts;
m_clocks.audioSetBufferEndPts(pts);
}
m_bytes_remaining = m_packet.packet.size;
m_packet_data = m_packet.packet.data;
}
else if (m_packet.type == FFmpegPacket::PACKET_END_OF_STREAM)
{
m_end_of_stream = true;
}
else if (m_packet.type == FFmpegPacket::PACKET_FLUSH)
{
avcodec_flush_buffers(m_context);
}
// just output silence when we reached the end of stream
if (m_end_of_stream)
{
memset(buffer, 0, size);
return size;
}
}
}
static void *adec_armdec_loop(void *args)
{
int ret;
int rlen = 0;
int inlen = 0;
int dlen = 0;
int declen = 0;
aml_audio_dec_t *audec;
audio_out_operations_t *aout_ops;
adec_cmd_t *msg = NULL;
AVPacket apkt;
char *inbuf = NULL;
char apkt_end[APACKET_END_SIZE];
char outbuf[AVCODEC_MAX_AUDIO_FRAME_SIZE];
int outlen = 0;
AVCodecContext *ctxCodec = NULL;
AVCodec *acodec = NULL;
int in_ape_fp = -1;
int out_ape_fp = -1;
int audio_handle = -1;
adec_print("adec_armdec_loop start!\n");
audec = (aml_audio_dec_t *)args;
aout_ops = &audec->aout_ops;
av_init_packet(&apkt);
//memset(inbuf, 0, READ_ABUFFER_SIZE);
memset(outbuf, 0, AVCODEC_MAX_AUDIO_FRAME_SIZE);
//buffer_stream_t init and set adsp_ops param
audec->bs=malloc(sizeof(buffer_stream_t));
int ret_value=init_buff(audec->bs);
if(ret_value==1)
adec_print("=====pcm buffer init ok buf_size:%d buf_data:0x%x end:0x%x !\n",audec->bs->buf_length,audec->bs->data,audec->bs->data+1024*1024);
audec->adsp_ops.dsp_on=1;
aout_ops->init(audec);
aout_ops->start(audec);
ctxCodec = avcodec_alloc_context();
if(!ctxCodec) {
adec_print("APE AVCodecContext allocate error!\n");
ctxCodec = NULL;
}
adec_print("ctxCodec!\n");
adec_print("adec_armdec_loop audec->pcodec = %d, audec->pcodec->ctxCodec = %d!\n", audec->pcodec, audec->pcodec->ctxCodec);
ctxCodec = audec->pcodec->ctxCodec;
ctxCodec->codec_type = CODEC_TYPE_AUDIO;
adec_print("open codec_id = %d--\n",ctxCodec->codec_id);
acodec = avcodec_find_decoder(ctxCodec->codec_id);
if (!acodec) {
adec_print("acodec not found\n");
}
adec_print("open codec_id = %d----------------------------------\n",ctxCodec->codec_id);
if (avcodec_open(ctxCodec, acodec) < 0) {
adec_print("Could not open acodec = %d\n", acodec);
}
out_ape_fp = open("./dump/123.dat", O_CREAT | O_RDWR);
if (out_ape_fp < 0) {
adec_print("Create input file failed! fd=%d------------------------------\n", out_ape_fp);
}
adec_print("out_ape_fp = %d!", out_ape_fp);
in_ape_fp = open("./dump/in.dat", O_CREAT | O_RDWR);
if (in_ape_fp < 0) {
adec_print("Create input file failed! fd=%d------------------------------\n", out_ape_fp);
}
adec_print("in_ape_fp = %d!", in_ape_fp);
ret = uio_init();
if (ret < 0){
adec_print("uio init error! \n");
goto error;
}
while (1){
if (inlen > 0) {
if (inbuf) {
free(inbuf);
inbuf = NULL;
}
inbuf = malloc(READ_ABUFFER_SIZE + inlen);
memcpy(inbuf, apkt_end, inlen);
rlen = read_buffer(inbuf+inlen, READ_ABUFFER_SIZE);
rlen += inlen;
}
else {
if (inbuf) {
free(inbuf);
inbuf = NULL;
}
inbuf = malloc(READ_ABUFFER_SIZE);
rlen = read_buffer(inbuf+inlen, READ_ABUFFER_SIZE);
}
if (out_ape_fp >= 0) {
write(in_ape_fp, inbuf, rlen);
adec_print("write ape data in rlen = %d bytes\n",rlen);
}
declen = 0;
if (rlen > 0){
inlen = rlen;
while (declen<rlen) {
outlen = AVCODEC_MAX_AUDIO_FRAME_SIZE;
dlen = decode_audio(ctxCodec, outbuf, &outlen, inbuf+declen, inlen);
if (dlen <= 0){
adec_print("dlen = %d error----\n",dlen);
if (inlen > 0) {
adec_print("packet end %d bytes----\n",inlen);
memcpy(apkt_end, (uint8_t *)(inbuf+declen), inlen);
}
break;
}
declen += dlen;
inlen -= dlen;
write_pcm_buffer(outbuf, audec->bs,outlen);
//write_buffer(outbuf, outlen);
if (outlen > 0) {
if (out_ape_fp >= 0) {
write(out_ape_fp, outbuf, outlen);
}
}
}
#if 0
outsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
apkt.data = inbuffer;
apkt.size = rlen;
while (apkt.size > 0){
dlen = avcodec_decode_audio3(ctxCodec, (int16_t *)outbuffer, &outsize, &apkt);
if (dlen <= 0){
adec_print("dlen = %d error----\n",dlen);
}
if (outsize > 0) {
if (out_ape_fp >= 0) {
write(out_ape_fp, outbuffer, outsize);
adec_print("write ape data%d bytes\n UUUUUUUU----\n",outsize);
}
}
declen += dlen;
apkt.size -= dlen;
if (apkt.size > 0){
apkt.data += dlen;
}
else if (apkt.size < 0){
adec_print("wrong aptk.size = %d, declen = %d, dlen = %d!", apkt.size, declen, dlen);
}
}
#endif
}
else {
adec_print("rlen = %d", rlen);
break;
}
}
close(out_ape_fp);
close(in_ape_fp);
adec_print("Exit adec_armdec_loop Thread!");
error:
pthread_exit(NULL);
return NULL;
}
static int init(sh_audio_t *sh_audio)
{
int tries = 0;
int x;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
AVDictionary *opts = NULL;
char tmpstr[50];
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
init_avcodec();
lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
return 0;
}
lavc_context = avcodec_alloc_context3(lavc_codec);
sh_audio->context=lavc_context;
snprintf(tmpstr, sizeof(tmpstr), "%f", drc_level);
av_dict_set(&opts, "drc_scale", tmpstr, 0);
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
if(sh_audio->wf){
lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->request_channels = audio_output_channels;
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
/* alloc extra data */
if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->wf->cbSize;
memcpy(lavc_context->extradata, sh_audio->wf + 1,
lavc_context->extradata_size);
}
// for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
{
lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
/* open it */
if (avcodec_open2(lavc_context, lavc_codec, &opts) < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
return 0;
}
av_dict_free(&opts);
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
// printf("\nFOURCC: 0x%X\n",sh_audio->format);
if(sh_audio->format==0x3343414D){
// MACE 3:1
sh_audio->ds->ss_div = 2*3; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
} else
if(sh_audio->format==0x3643414D){
// MACE 6:1
sh_audio->ds->ss_div = 2*6; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
}
// Decode at least 1 byte: (to get header filled)
do {
x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
} while (x <= 0 && tries++ < 5);
if(x>0) sh_audio->a_buffer_len=x;
sh_audio->i_bps=lavc_context->bit_rate/8;
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
switch (lavc_context->sample_fmt) {
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S32:
case AV_SAMPLE_FMT_FLT:
break;
default:
return 0;
}
return 1;
}
/* pkt = NULL means EOF (needed to flush decoder buffers) */
static int output_packet(InputStream *ist, const AVPacket *pkt)
{
int ret = 0, i;
int got_output = 0;
AVPacket avpkt;
if (!ist->saw_first_ts) {
ist->dts = ist->st->avg_frame_rate.num ? - ist->dec_ctx->has_b_frames * AV_TIME_BASE / av_q2d(ist->st->avg_frame_rate) : 0;
ist->pts = 0;
if (pkt != NULL && pkt->pts != AV_NOPTS_VALUE && !ist->decoding_needed) {
ist->dts += av_rescale_q(pkt->pts, ist->st->time_base, AV_TIME_BASE_Q);
ist->pts = ist->dts; //unused but better to set it to a value thats not totally wrong
}
ist->saw_first_ts = 1;
}
if (ist->next_dts == AV_NOPTS_VALUE)
ist->next_dts = ist->dts;
if (ist->next_pts == AV_NOPTS_VALUE)
ist->next_pts = ist->pts;
if (pkt == NULL) {
/* EOF handling */
av_init_packet(&avpkt);
avpkt.data = NULL;
avpkt.size = 0;
goto handle_eof;
} else {
avpkt = *pkt;
}
if (pkt->dts != AV_NOPTS_VALUE) {
ist->next_dts = ist->dts = av_rescale_q(pkt->dts, ist->st->time_base, AV_TIME_BASE_Q);
if (ist->dec_ctx->codec_type != AVMEDIA_TYPE_VIDEO || !ist->decoding_needed)
ist->next_pts = ist->pts = ist->dts;
}
// while we have more to decode or while the decoder did output something on EOF
while (ist->decoding_needed && (avpkt.size > 0 || (!pkt && got_output))) {
int duration;
handle_eof:
ist->pts = ist->next_pts;
ist->dts = ist->next_dts;
if (avpkt.size && avpkt.size != pkt->size &&
!(ist->dec->capabilities & CODEC_CAP_SUBFRAMES)) {
av_log(NULL, ist->showed_multi_packet_warning ? AV_LOG_VERBOSE : AV_LOG_WARNING,
"Multiple frames in a packet from stream %d\n", pkt->stream_index);
ist->showed_multi_packet_warning = 1;
}
switch (ist->dec_ctx->codec_type) {
case AVMEDIA_TYPE_AUDIO:
ret = decode_audio (ist, &avpkt, &got_output);
break;
case AVMEDIA_TYPE_VIDEO:
ret = decode_video (ist, &avpkt, &got_output);
if (avpkt.duration) {
duration = av_rescale_q(avpkt.duration, ist->st->time_base, AV_TIME_BASE_Q);
} else if(ist->dec_ctx->time_base.num != 0 && ist->dec_ctx->time_base.den != 0) {
int ticks= ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->dec_ctx->ticks_per_frame;
duration = ((int64_t)AV_TIME_BASE *
ist->dec_ctx->time_base.num * ticks) /
ist->dec_ctx->time_base.den;
} else
duration = 0;
if(ist->dts != AV_NOPTS_VALUE && duration) {
ist->next_dts += duration;
}else
ist->next_dts = AV_NOPTS_VALUE;
if (got_output)
ist->next_pts += duration; //FIXME the duration is not correct in some cases
break;
case AVMEDIA_TYPE_SUBTITLE:
ret = transcode_subtitles(ist, &avpkt, &got_output);
break;
default:
return -1;
}
if (ret < 0)
return ret;
avpkt.dts=
avpkt.pts= AV_NOPTS_VALUE;
// touch data and size only if not EOF
if (pkt) {
if(ist->dec_ctx->codec_type != AVMEDIA_TYPE_AUDIO)
ret = avpkt.size;
avpkt.data += ret;
avpkt.size -= ret;
}
if (!got_output) {
continue;
}
}
/* handle stream copy */
if (!ist->decoding_needed) {
ist->dts = ist->next_dts;
switch (ist->dec_ctx->codec_type) {
case AVMEDIA_TYPE_AUDIO:
ist->next_dts += ((int64_t)AV_TIME_BASE * ist->dec_ctx->frame_size) /
ist->dec_ctx->sample_rate;
break;
case AVMEDIA_TYPE_VIDEO:
if (ist->framerate.num) {
// TODO: Remove work-around for c99-to-c89 issue 7
AVRational time_base_q = AV_TIME_BASE_Q;
int64_t next_dts = av_rescale_q(ist->next_dts, time_base_q, av_inv_q(ist->framerate));
ist->next_dts = av_rescale_q(next_dts + 1, av_inv_q(ist->framerate), time_base_q);
} else if (pkt->duration) {
ist->next_dts += av_rescale_q(pkt->duration, ist->st->time_base, AV_TIME_BASE_Q);
} else if(ist->dec_ctx->time_base.num != 0) {
int ticks= ist->st->parser ? ist->st->parser->repeat_pict + 1 : ist->dec_ctx->ticks_per_frame;
ist->next_dts += ((int64_t)AV_TIME_BASE *
ist->dec_ctx->time_base.num * ticks) /
ist->dec_ctx->time_base.den;
}
break;
}
ist->pts = ist->dts;
ist->next_pts = ist->next_dts;
}
for (i = 0; pkt && i < nb_output_streams; i++) {
OutputStream *ost = output_streams[i];
if (!check_output_constraints(ist, ost) || ost->encoding_needed)
continue;
do_streamcopy(ist, ost, pkt);
}
return 0;
}
