int ff_mov_add_hinted_packet(AVFormatContext *s, AVPacket *pkt,
int track_index, int sample)
{
MOVMuxContext *mov = s->priv_data;
MOVTrack *trk = &mov->tracks[track_index];
AVFormatContext *rtp_ctx = trk->rtp_ctx;
uint8_t *buf = NULL;
int size;
AVIOContext *hintbuf = NULL;
AVPacket hint_pkt;
int ret = 0, count;
if (!rtp_ctx)
return AVERROR(ENOENT);
if (!rtp_ctx->pb)
return AVERROR(ENOMEM);
sample_queue_push(&trk->sample_queue, pkt, sample);
/* Feed the packet to the RTP muxer */
ff_write_chained(rtp_ctx, 0, pkt, s);
/* Fetch the output from the RTP muxer, open a new output buffer
* for next time. */
size = url_close_dyn_buf(rtp_ctx->pb, &buf);
if ((ret = url_open_dyn_packet_buf(&rtp_ctx->pb,
RTP_MAX_PACKET_SIZE)) < 0)
goto done;
if (size <= 0)
goto done;
/* Open a buffer for writing the hint */
if ((ret = url_open_dyn_buf(&hintbuf)) < 0)
goto done;
av_init_packet(&hint_pkt);
count = write_hint_packets(hintbuf, buf, size, trk, &hint_pkt.dts);
av_freep(&buf);
/* Write the hint data into the hint track */
hint_pkt.size = size = url_close_dyn_buf(hintbuf, &buf);
hint_pkt.data = buf;
hint_pkt.pts = hint_pkt.dts;
hint_pkt.stream_index = track_index;
if (pkt->flags & AV_PKT_FLAG_KEY)
hint_pkt.flags |= AV_PKT_FLAG_KEY;
if (count > 0)
ff_mov_write_packet(s, &hint_pkt);
done:
av_free(buf);
sample_queue_retain(&trk->sample_queue);
return ret;
}
static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
AVFormatContext *rtpctx = rtsp_st->transport_priv;
uint8_t *buf, *ptr;
int size;
uint8_t interleave_header[4];
size = url_close_dyn_buf(rtpctx->pb, &buf);
ptr = buf;
while (size > 4) {
uint32_t packet_len = AV_RB32(ptr);
int id;
ptr += 4;
size -= 4;
if (packet_len > size || packet_len < 2)
break;
if (ptr[1] >= 200 && ptr[1] <= 204)
id = rtsp_st->interleaved_max; /* RTCP */
else
id = rtsp_st->interleaved_min; /* RTP */
interleave_header[0] = '$';
interleave_header[1] = id;
AV_WB16(interleave_header + 2, packet_len);
url_write(rt->rtsp_hd, interleave_header, 4);
url_write(rt->rtsp_hd, ptr, packet_len);
ptr += packet_len;
size -= packet_len;
}
av_free(buf);
url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
return 0;
}
AVFormatContext *ff_rtp_chain_mux_open(AVFormatContext *s, AVStream *st,
URLContext *handle, int packet_size)
{
AVFormatContext *rtpctx;
int ret;
AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
if (!rtp_format)
return NULL;
/* Allocate an AVFormatContext for each output stream */
rtpctx = avformat_alloc_context();
if (!rtpctx)
return NULL;
rtpctx->oformat = rtp_format;
if (!av_new_stream(rtpctx, 0)) {
av_free(rtpctx);
return NULL;
}
/* Copy the max delay setting; the rtp muxer reads this. */
rtpctx->max_delay = s->max_delay;
/* Copy other stream parameters. */
rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
/* Set the synchronized start time. */
rtpctx->start_time_realtime = s->start_time_realtime;
/* Remove the local codec, link to the original codec
* context instead, to give the rtp muxer access to
* codec parameters. */
av_free(rtpctx->streams[0]->codec);
rtpctx->streams[0]->codec = st->codec;
if (handle) {
url_fdopen(&rtpctx->pb, handle);
} else
url_open_dyn_packet_buf(&rtpctx->pb, packet_size);
ret = av_write_header(rtpctx);
if (ret) {
if (handle) {
url_fclose(rtpctx->pb);
} else {
uint8_t *ptr;
url_close_dyn_buf(rtpctx->pb, &ptr);
av_free(ptr);
}
av_free(rtpctx->streams[0]);
av_free(rtpctx);
return NULL;
}
/* Copy the RTP AVStream timebase back to the original AVStream */
st->time_base = rtpctx->streams[0]->time_base;
return rtpctx;
}
static void svq3_extradata_free(PayloadContext *sv)
{
if (sv->pktbuf) {
uint8_t *buf;
url_close_dyn_buf(sv->pktbuf, &buf);
av_free(buf);
}
av_free(sv);
}
static void prepare_packet(AVPacket *pkt, PayloadContext *vp8, int stream)
{
av_init_packet(pkt);
pkt->stream_index = stream;
pkt->flags = vp8->is_keyframe ? AV_PKT_FLAG_KEY : 0;
pkt->size = url_close_dyn_buf(vp8->data, &pkt->data);
pkt->destruct = av_destruct_packet;
vp8->data = NULL;
}
static void vp8_free_context(PayloadContext *vp8)
{
if (vp8->data) {
uint8_t *tmp;
url_close_dyn_buf(vp8->data, &tmp);
av_free(tmp);
}
av_free(vp8);
}
static void asfrtp_free_context(PayloadContext *asf)
{
if (asf->pktbuf) {
uint8_t *p = NULL;
url_close_dyn_buf(asf->pktbuf, &p);
asf->pktbuf = NULL;
av_free(p);
}
av_freep(&asf->buf);
av_free(asf);
}
int ff_avc_parse_nal_units_buf(const uint8_t *buf_in, uint8_t **buf, int *size)
{
ByteIOContext *pb;
int ret = url_open_dyn_buf(&pb);
if(ret < 0)
return ret;
ff_avc_parse_nal_units(pb, buf_in, *size);
av_freep(buf);
*size = url_close_dyn_buf(pb, buf);
return 0;
}
static void put_str16(ByteIOContext *s, const char *tag)
{
int len;
uint8_t *pb;
ByteIOContext *dyn_buf;
if (url_open_dyn_buf(&dyn_buf) < 0)
return;
avio_put_str16le(dyn_buf, tag);
len = url_close_dyn_buf(dyn_buf, &pb);
put_le16(s, len);
put_buffer(s, pb, len);
av_freep(&pb);
}
void rtp_send_punch_packets(URLContext* rtp_handle)
{
AVIOContext *pb;
uint8_t *buf;
int len;
/* Send a small RTP packet */
if (url_open_dyn_buf(&pb) < 0)
return;
avio_w8(pb, (RTP_VERSION << 6));
avio_w8(pb, 0); /* Payload type */
avio_wb16(pb, 0); /* Seq */
avio_wb32(pb, 0); /* Timestamp */
avio_wb32(pb, 0); /* SSRC */
avio_flush(pb);
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
url_write(rtp_handle, buf, len);
av_free(buf);
/* Send a minimal RTCP RR */
if (url_open_dyn_buf(&pb) < 0)
return;
avio_w8(pb, (RTP_VERSION << 6));
avio_w8(pb, RTCP_RR); /* receiver report */
avio_wb16(pb, 1); /* length in words - 1 */
avio_wb32(pb, 0); /* our own SSRC */
avio_flush(pb);
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
url_write(rtp_handle, buf, len);
av_free(buf);
}
void ff_mov_close_hinting(MOVTrack *track) {
AVFormatContext* rtp_ctx = track->rtp_ctx;
uint8_t *ptr;
av_freep(&track->enc);
sample_queue_free(&track->sample_queue);
if (!rtp_ctx)
return;
if (rtp_ctx->pb) {
av_write_trailer(rtp_ctx);
url_close_dyn_buf(rtp_ctx->pb, &ptr);
av_free(ptr);
}
avformat_free_context(rtp_ctx);
}
int ff_avc_parse_nal_units(const uint8_t *buf_in, uint8_t **buf, int *size)
{
ByteIOContext pb;
const uint8_t *p = buf_in;
const uint8_t *end = p + *size;
const uint8_t *nal_start, *nal_end;
int ret = url_open_dyn_buf(&pb);
if(ret < 0)
return ret;
nal_start = ff_avc_find_startcode(p, end);
while (nal_start < end) {
while(!*(nal_start++));
nal_end = ff_avc_find_startcode(nal_start, end);
put_be32(&pb, nal_end - nal_start);
put_buffer(&pb, nal_start, nal_end - nal_start);
nal_start = nal_end;
}
av_freep(buf);
*size = url_close_dyn_buf(&pb, buf);
return 0;
}
static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
AVFormatContext *rtpctx = rtsp_st->transport_priv;
uint8_t *buf, *ptr;
int size;
uint8_t *interleave_header, *interleaved_packet;
size = url_close_dyn_buf(rtpctx->pb, &buf);
ptr = buf;
while (size > 4) {
uint32_t packet_len = AV_RB32(ptr);
int id;
/* The interleaving header is exactly 4 bytes, which happens to be
* the same size as the packet length header from
* url_open_dyn_packet_buf. So by writing the interleaving header
* over these bytes, we get a consecutive interleaved packet
* that can be written in one call. */
interleaved_packet = interleave_header = ptr;
ptr += 4;
size -= 4;
if (packet_len > size || packet_len < 2)
break;
if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
id = rtsp_st->interleaved_max; /* RTCP */
else
id = rtsp_st->interleaved_min; /* RTP */
interleave_header[0] = '$';
interleave_header[1] = id;
AV_WB16(interleave_header + 2, packet_len);
url_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
ptr += packet_len;
size -= packet_len;
}
av_free(buf);
url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
return 0;
}
static int ogg_write_page(AVFormatContext *s, OGGPage *page, int extra_flags)
{
OGGStreamContext *oggstream = s->streams[page->stream_index]->priv_data;
ByteIOContext *pb;
int64_t crc_offset;
int ret, size;
uint8_t *buf;
ret = url_open_dyn_buf(&pb);
if (ret < 0)
return ret;
init_checksum(pb, ff_crc04C11DB7_update, 0);
put_tag(pb, "OggS");
put_byte(pb, 0);
put_byte(pb, page->flags | extra_flags);
put_le64(pb, page->granule);
put_le32(pb, oggstream->serial_num);
put_le32(pb, oggstream->page_counter++);
crc_offset = url_ftell(pb);
put_le32(pb, 0); // crc
put_byte(pb, page->segments_count);
put_buffer(pb, page->segments, page->segments_count);
put_buffer(pb, page->data, page->size);
ogg_update_checksum(s, pb, crc_offset);
put_flush_packet(pb);
size = url_close_dyn_buf(pb, &buf);
if (size < 0)
return size;
put_buffer(s->pb, buf, size);
put_flush_packet(s->pb);
av_free(buf);
oggstream->page_count--;
return 0;
}
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
AVIOContext *pb;
uint8_t *buf;
int len;
int rtcp_bytes;
RTPStatistics *stats= &s->statistics;
uint32_t lost;
uint32_t extended_max;
uint32_t expected_interval;
uint32_t received_interval;
uint32_t lost_interval;
uint32_t expected;
uint32_t fraction;
uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
if (!s->rtp_ctx || (count < 1))
return -1;
/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
s->octet_count += count;
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN;
rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
if (rtcp_bytes < 28)
return -1;
s->last_octet_count = s->octet_count;
if (url_open_dyn_buf(&pb) < 0)
return -1;
// Receiver Report
avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
avio_w8(pb, RTCP_RR);
avio_wb16(pb, 7); /* length in words - 1 */
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
avio_wb32(pb, s->ssrc + 1);
avio_wb32(pb, s->ssrc); // server SSRC
// some placeholders we should really fill...
// RFC 1889/p64
extended_max= stats->cycles + stats->max_seq;
expected= extended_max - stats->base_seq + 1;
lost= expected - stats->received;
lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
expected_interval= expected - stats->expected_prior;
stats->expected_prior= expected;
received_interval= stats->received - stats->received_prior;
stats->received_prior= stats->received;
lost_interval= expected_interval - received_interval;
if (expected_interval==0 || lost_interval<=0) fraction= 0;
else fraction = (lost_interval<<8)/expected_interval;
fraction= (fraction<<24) | lost;
avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
avio_wb32(pb, extended_max); /* max sequence received */
avio_wb32(pb, stats->jitter>>4); /* jitter */
if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
{
avio_wb32(pb, 0); /* last SR timestamp */
avio_wb32(pb, 0); /* delay since last SR */
} else {
uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
avio_wb32(pb, middle_32_bits); /* last SR timestamp */
avio_wb32(pb, delay_since_last); /* delay since last SR */
}
// CNAME
avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
avio_w8(pb, RTCP_SDES);
len = strlen(s->hostname);
avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
avio_wb32(pb, s->ssrc);
avio_w8(pb, 0x01);
avio_w8(pb, len);
avio_write(pb, s->hostname, len);
// padding
for (len = (6 + len) % 4; len % 4; len++) {
avio_w8(pb, 0);
}
avio_flush(pb);
len = url_close_dyn_buf(pb, &buf);
if ((len > 0) && buf) {
int result;
av_dlog(s->ic, "sending %d bytes of RR\n", len);
result= url_write(s->rtp_ctx, buf, len);
av_dlog(s->ic, "result from url_write: %d\n", result);
av_free(buf);
}
return 0;
}
/** return 0 on packet, <0 on partial packet or error... */
static int svq3_parse_packet (AVFormatContext *s, PayloadContext *sv,
AVStream *st, AVPacket *pkt,
uint32_t *timestamp,
const uint8_t *buf, int len, int flags)
{
int config_packet, start_packet, end_packet;
if (len < 2)
return AVERROR_INVALIDDATA;
config_packet = buf[0] & 0x40;
start_packet = buf[0] & 0x20;
end_packet = buf[0] & 0x10;
buf += 2; // ignore buf[1]
len -= 2;
if (config_packet) {
av_freep(&st->codec->extradata);
st->codec->extradata_size = 0;
if (len < 2 || !(st->codec->extradata =
av_malloc(len + 8 + FF_INPUT_BUFFER_PADDING_SIZE)))
return AVERROR_INVALIDDATA;
st->codec->extradata_size = len + 8;
memcpy(st->codec->extradata, "SEQH", 4);
AV_WB32(st->codec->extradata + 4, len);
memcpy(st->codec->extradata + 8, buf, len);
/* We set codec_id to CODEC_ID_NONE initially to
* delay decoder initialization since extradata is
* carried within the RTP stream, not SDP. Here,
* by setting codec_id to CODEC_ID_SVQ3, we are signalling
* to the decoder that it is OK to initialize. */
st->codec->codec_id = CODEC_ID_SVQ3;
return AVERROR(EAGAIN);
}
if (start_packet) {
int res;
if (sv->pktbuf) {
uint8_t *tmp;
url_close_dyn_buf(sv->pktbuf, &tmp);
av_free(tmp);
}
if ((res = url_open_dyn_buf(&sv->pktbuf)) < 0)
return res;
sv->timestamp = *timestamp;
}
if (!sv->pktbuf)
return AVERROR_INVALIDDATA;
put_buffer(sv->pktbuf, buf, len);
if (end_packet) {
av_init_packet(pkt);
pkt->stream_index = st->index;
*timestamp = sv->timestamp;
pkt->size = url_close_dyn_buf(sv->pktbuf, &pkt->data);
pkt->destruct = av_destruct_packet;
sv->pktbuf = NULL;
return 0;
}
return AVERROR(EAGAIN);
}
/**
* @return 0 when a packet was written into /p pkt, and no more data is left;
* 1 when a packet was written into /p pkt, and more packets might be left;
* <0 when not enough data was provided to return a full packet, or on error.
*/
static int asfrtp_parse_packet(AVFormatContext *s, PayloadContext *asf,
AVStream *st, AVPacket *pkt,
uint32_t *timestamp,
const uint8_t *buf, int len, int flags)
{
ByteIOContext *pb = &asf->pb;
int res, mflags, len_off;
RTSPState *rt = s->priv_data;
if (!rt->asf_ctx)
return -1;
if (len > 0) {
int off, out_len = 0;
if (len < 4)
return -1;
av_freep(&asf->buf);
init_put_byte(pb, buf, len, 0, NULL, NULL, NULL, NULL);
while (url_ftell(pb) + 4 < len) {
int start_off = url_ftell(pb);
mflags = get_byte(pb);
if (mflags & 0x80)
flags |= RTP_FLAG_KEY;
len_off = get_be24(pb);
if (mflags & 0x20) /**< relative timestamp */
url_fskip(pb, 4);
if (mflags & 0x10) /**< has duration */
url_fskip(pb, 4);
if (mflags & 0x8) /**< has location ID */
url_fskip(pb, 4);
off = url_ftell(pb);
if (!(mflags & 0x40)) {
/**
* If 0x40 is not set, the len_off field specifies an offset
* of this packet's payload data in the complete (reassembled)
* ASF packet. This is used to spread one ASF packet over
* multiple RTP packets.
*/
if (asf->pktbuf && len_off != url_ftell(asf->pktbuf)) {
uint8_t *p;
url_close_dyn_buf(asf->pktbuf, &p);
asf->pktbuf = NULL;
av_free(p);
}
if (!len_off && !asf->pktbuf &&
(res = url_open_dyn_buf(&asf->pktbuf)) < 0)
return res;
if (!asf->pktbuf)
return AVERROR(EIO);
put_buffer(asf->pktbuf, buf + off, len - off);
url_fskip(pb, len - off);
if (!(flags & RTP_FLAG_MARKER))
return -1;
out_len = url_close_dyn_buf(asf->pktbuf, &asf->buf);
asf->pktbuf = NULL;
} else {
/**
* If 0x40 is set, the len_off field specifies the length of
* the next ASF packet that can be read from this payload
* data alone. This is commonly the same as the payload size,
* but could be less in case of packet splitting (i.e.
* multiple ASF packets in one RTP packet).
*/
int cur_len = start_off + len_off - off;
int prev_len = out_len;
out_len += cur_len;
asf->buf = av_realloc(asf->buf, out_len);
memcpy(asf->buf + prev_len, buf + off,
FFMIN(cur_len, len - off));
url_fskip(pb, cur_len);
}
}
init_packetizer(pb, asf->buf, out_len);
pb->pos += rt->asf_pb_pos;
pb->eof_reached = 0;
rt->asf_ctx->pb = pb;
}
for (;;) {
int i;
res = av_read_packet(rt->asf_ctx, pkt);
rt->asf_pb_pos = url_ftell(pb);
if (res != 0)
break;
for (i = 0; i < s->nb_streams; i++) {
if (s->streams[i]->id == rt->asf_ctx->streams[pkt->stream_index]->id) {
pkt->stream_index = i;
return 1; // FIXME: return 0 if last packet
}
}
av_free_packet(pkt);
}
return res == 1 ? -1 : res;
}
/* XXX: This class should really pass the picture parameters by a separate API
* so that we can, by contract, enforce that the frame size can't suddenly
* change on us. */
void Java_org_devtcg_rojocam_ffmpeg_RtpOutputContext_nativeWriteFrame(JNIEnv *env,
jclass clazz, jint nativeInt, jbyteArray data, jlong frameTime,
jint frameFormat, jint frameWidth, jint frameHeight,
jint frameBitsPerPixel) {
RtpOutputContext *rtpContext = (RtpOutputContext *)nativeInt;
AVFormatContext *avContext;
AVCodecContext *codec;
AVStream *outputStream;
AVPacket pkt;
jbyte *data_c;
int max_packet_size;
uint8_t *rtp_data;
int rtp_data_len;
avContext = rtpContext->avContext;
outputStream = avContext->streams[0];
codec = outputStream->codec;
if (rtpContext->tempFrame == NULL) {
if (!first_frame_init(env, rtpContext, frameFormat,
frameWidth, frameHeight)) {
LOGE("Error initializing encoding buffers, cannot stream");
return;
}
}
data_c = (*env)->GetByteArrayElements(env, data, NULL);
/* Convert the input arguments to an AVPacket, simulating it as though we
* read this from the ffmpeg libraries but there was no need to do this as
* it was passed into us already as a raw video frame. */
int frameDuration = frameTime - rtpContext->lastFrameTime;
bool frameEncoded = encode_video_frame(rtpContext,
outputStream, rtpContext->tempFrame, rtpContext->imgConvert,
rtpContext->tempEncodedBuf, sizeof(rtpContext->tempEncodedBuf),
data_c, frameTime, frameDuration, frameFormat,
frameWidth, frameHeight, frameBitsPerPixel, &pkt);
rtpContext->lastFrameTime = frameTime;
(*env)->ReleaseByteArrayElements(env, data, data_c, JNI_ABORT);
if (frameEncoded) {
#if PROFILE_WRITE_FRAME
struct timeval then;
gettimeofday(&then, NULL);
#endif
max_packet_size = url_get_max_packet_size(rtpContext->urlContext);
url_open_dyn_packet_buf(&avContext->pb, max_packet_size);
avContext->pb->seekable = 0;
/* This organizes our encoded packet into RTP packet segments (but it
* doesn't actually send anything over the network yet). */
if (av_write_frame(avContext, &pkt) < 0) {
jniThrowException(env, "java/io/IOException", "Error writing frame to output");
}
/* Actually deliver the packetized RTP data to the remote peer. */
rtp_data_len = url_close_dyn_buf(avContext->pb, &rtp_data);
exhaustive_send(rtpContext->urlContext, rtp_data, rtp_data_len);
av_free(rtp_data);
/* XXX: I dunno, ffserver.c does this... */
outputStream->codec->frame_number++;
#if PROFILE_WRITE_FRAME
store_elapsed(&rtpContext->write_time, &then);
#endif
} else {
#if PROFILE_WRITE_TIME
rtpContext->write_time = 0;
#endif
}
#if PROFILE_WRITE_FRAME
//LOGI("resample@%ld ms; encode@%ld ms; write@%ld ms",
// rtpContext->resampling_time, rtpContext->encoding_time,
// rtpContext->write_time);
#endif
}
jint Java_org_devtcg_rojocam_ffmpeg_RtpOutputContext_nativeCreate(JNIEnv *env,
jclass clazz, jint streamConfigNativeInt, jlong nowNanoTime,
jstring hostAddress, jint rtpPort) {
FFStreamConfig *config = (FFStreamConfig *)streamConfigNativeInt;
RtpOutputContext *rtpContext = NULL;
AVFormatContext *avContext = NULL;
AVStream *st = NULL;
uint8_t *dummy_buf;
int max_packet_size;
rtpContext = av_mallocz(sizeof(RtpOutputContext));
if (rtpContext == NULL) {
jniThrowOOM(env);
goto cleanup;
}
rtpContext->lastFrameTime = nowNanoTime;
rtpContext->config = config;
avContext = avformat_alloc_context();
if (avContext == NULL) {
jniThrowOOM(env);
goto cleanup;
}
avContext->oformat = av_guess_format("rtp", NULL, NULL);
if (avContext->oformat == NULL) {
jniThrowException(env, "java/lang/IllegalStateException",
"rtp avformat is not available");
goto cleanup;
}
rtpContext->avContext = avContext;
st = av_mallocz(sizeof(AVStream));
if (st == NULL) {
jniThrowOOM(env);
goto cleanup;
}
avContext->nb_streams = 1;
avContext->streams = av_malloc(avContext->nb_streams * sizeof(*avContext->streams));
avContext->streams[0] = st;
/* XXX: What would we be doing if we supported audio as well? */
memcpy(st, config->streams[0], sizeof(AVStream));
st->priv_data = NULL;
const jbyte *hostAddress_str = (*env)->GetStringUTFChars(env,
hostAddress, NULL);
snprintf(avContext->filename, sizeof(avContext->filename),
"rtp://%s:%d?localrtpport=5000&localrtcpport=5001",
hostAddress_str, rtpPort);
(*env)->ReleaseStringUTFChars(env, hostAddress, hostAddress_str);
if (url_open(&rtpContext->urlContext,
avContext->filename, URL_WRONLY) < 0) {
LOGE("Cannot open url context for filename=%s", avContext->filename);
jniThrowException(env, "java/io/IOException", "Unable to open URL");
goto cleanup;
}
max_packet_size = url_get_max_packet_size(rtpContext->urlContext);
/* XXX: No idea what purpose this serves... */
url_open_dyn_packet_buf(&avContext->pb, max_packet_size);
av_set_parameters(avContext, NULL);
if (av_write_header(avContext) < 0) {
jniThrowException(env, "java/io/IOException", "Unexpected error writing dummy RTP header");
goto cleanup;
}
url_close_dyn_buf(avContext->pb, &dummy_buf);
av_free(dummy_buf);
return (jint)rtpContext;
cleanup:
rtp_output_context_free(rtpContext);
assert((*env)->ExceptionOccurred(env));
return 0;
}
/* write the header (used two times if non streamed) */
static int asf_write_header1(AVFormatContext *s, int64_t file_size, int64_t data_chunk_size)
{
ASFContext *asf = s->priv_data;
ByteIOContext *pb = s->pb;
AVMetadataTag *tags[5];
int header_size, n, extra_size, extra_size2, wav_extra_size, file_time;
int has_title;
int metadata_count;
AVCodecContext *enc;
int64_t header_offset, cur_pos, hpos;
int bit_rate;
int64_t duration;
ff_metadata_conv(&s->metadata, ff_asf_metadata_conv, NULL);
tags[0] = av_metadata_get(s->metadata, "title" , NULL, 0);
tags[1] = av_metadata_get(s->metadata, "author" , NULL, 0);
tags[2] = av_metadata_get(s->metadata, "copyright", NULL, 0);
tags[3] = av_metadata_get(s->metadata, "comment" , NULL, 0);
tags[4] = av_metadata_get(s->metadata, "rating" , NULL, 0);
duration = asf->duration + PREROLL_TIME * 10000;
has_title = tags[0] || tags[1] || tags[2] || tags[3] || tags[4];
metadata_count = s->metadata ? s->metadata->count : 0;
bit_rate = 0;
for(n=0;n<s->nb_streams;n++) {
enc = s->streams[n]->codec;
av_set_pts_info(s->streams[n], 32, 1, 1000); /* 32 bit pts in ms */
bit_rate += enc->bit_rate;
}
if (asf->is_streamed) {
put_chunk(s, 0x4824, 0, 0xc00); /* start of stream (length will be patched later) */
}
put_guid(pb, &ff_asf_header);
put_le64(pb, -1); /* header length, will be patched after */
put_le32(pb, 3 + has_title + !!metadata_count + s->nb_streams); /* number of chunks in header */
put_byte(pb, 1); /* ??? */
put_byte(pb, 2); /* ??? */
/* file header */
header_offset = url_ftell(pb);
hpos = put_header(pb, &ff_asf_file_header);
put_guid(pb, &ff_asf_my_guid);
put_le64(pb, file_size);
file_time = 0;
put_le64(pb, unix_to_file_time(file_time));
put_le64(pb, asf->nb_packets); /* number of packets */
put_le64(pb, duration); /* end time stamp (in 100ns units) */
put_le64(pb, asf->duration); /* duration (in 100ns units) */
put_le64(pb, PREROLL_TIME); /* start time stamp */
put_le32(pb, (asf->is_streamed || url_is_streamed(pb)) ? 3 : 2); /* ??? */
put_le32(pb, s->packet_size); /* packet size */
put_le32(pb, s->packet_size); /* packet size */
put_le32(pb, bit_rate); /* Nominal data rate in bps */
end_header(pb, hpos);
/* unknown headers */
hpos = put_header(pb, &ff_asf_head1_guid);
put_guid(pb, &ff_asf_head2_guid);
put_le32(pb, 6);
put_le16(pb, 0);
end_header(pb, hpos);
/* title and other infos */
if (has_title) {
int len;
uint8_t *buf;
ByteIOContext *dyn_buf;
if (url_open_dyn_buf(&dyn_buf) < 0)
return AVERROR(ENOMEM);
hpos = put_header(pb, &ff_asf_comment_header);
for (n = 0; n < FF_ARRAY_ELEMS(tags); n++) {
len = tags[n] ? avio_put_str16le(dyn_buf, tags[n]->value) : 0;
put_le16(pb, len);
}
len = url_close_dyn_buf(dyn_buf, &buf);
put_buffer(pb, buf, len);
av_freep(&buf);
end_header(pb, hpos);
}
if (metadata_count) {
AVMetadataTag *tag = NULL;
hpos = put_header(pb, &ff_asf_extended_content_header);
put_le16(pb, metadata_count);
while ((tag = av_metadata_get(s->metadata, "", tag, AV_METADATA_IGNORE_SUFFIX))) {
put_str16(pb, tag->key);
put_le16(pb, 0);
put_str16(pb, tag->value);
}
end_header(pb, hpos);
}
/* stream headers */
for(n=0;n<s->nb_streams;n++) {
int64_t es_pos;
// ASFStream *stream = &asf->streams[n];
enc = s->streams[n]->codec;
asf->streams[n].num = n + 1;
asf->streams[n].seq = 0;
switch(enc->codec_type) {
case AVMEDIA_TYPE_AUDIO:
wav_extra_size = 0;
extra_size = 18 + wav_extra_size;
extra_size2 = 8;
break;
default:
case AVMEDIA_TYPE_VIDEO:
wav_extra_size = enc->extradata_size;
extra_size = 0x33 + wav_extra_size;
extra_size2 = 0;
break;
}
hpos = put_header(pb, &ff_asf_stream_header);
if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {
put_guid(pb, &ff_asf_audio_stream);
put_guid(pb, &ff_asf_audio_conceal_spread);
} else {
put_guid(pb, &ff_asf_video_stream);
put_guid(pb, &ff_asf_video_conceal_none);
}
put_le64(pb, 0); /* ??? */
es_pos = url_ftell(pb);
put_le32(pb, extra_size); /* wav header len */
put_le32(pb, extra_size2); /* additional data len */
put_le16(pb, n + 1); /* stream number */
put_le32(pb, 0); /* ??? */
if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {
/* WAVEFORMATEX header */
int wavsize = ff_put_wav_header(pb, enc);
if ((enc->codec_id != CODEC_ID_MP3) && (enc->codec_id != CODEC_ID_MP2) && (enc->codec_id != CODEC_ID_ADPCM_IMA_WAV) && (enc->extradata_size==0)) {
wavsize += 2;
put_le16(pb, 0);
}
if (wavsize < 0)
return -1;
if (wavsize != extra_size) {
cur_pos = url_ftell(pb);
url_fseek(pb, es_pos, SEEK_SET);
put_le32(pb, wavsize); /* wav header len */
url_fseek(pb, cur_pos, SEEK_SET);
}
/* ERROR Correction */
put_byte(pb, 0x01);
if(enc->codec_id == CODEC_ID_ADPCM_G726 || !enc->block_align){
put_le16(pb, 0x0190);
put_le16(pb, 0x0190);
}else{
put_le16(pb, enc->block_align);
put_le16(pb, enc->block_align);
}
put_le16(pb, 0x01);
put_byte(pb, 0x00);
} else {
put_le32(pb, enc->width);
put_le32(pb, enc->height);
put_byte(pb, 2); /* ??? */
put_le16(pb, 40 + enc->extradata_size); /* size */
/* BITMAPINFOHEADER header */
ff_put_bmp_header(pb, enc, ff_codec_bmp_tags, 1);
}
end_header(pb, hpos);
}
/* media comments */
hpos = put_header(pb, &ff_asf_codec_comment_header);
put_guid(pb, &ff_asf_codec_comment1_header);
put_le32(pb, s->nb_streams);
for(n=0;n<s->nb_streams;n++) {
AVCodec *p;
const char *desc;
int len;
uint8_t *buf;
ByteIOContext *dyn_buf;
enc = s->streams[n]->codec;
p = avcodec_find_encoder(enc->codec_id);
if(enc->codec_type == AVMEDIA_TYPE_AUDIO)
put_le16(pb, 2);
else if(enc->codec_type == AVMEDIA_TYPE_VIDEO)
put_le16(pb, 1);
else
put_le16(pb, -1);
if(enc->codec_id == CODEC_ID_WMAV2)
desc = "Windows Media Audio V8";
else
desc = p ? p->name : enc->codec_name;
if ( url_open_dyn_buf(&dyn_buf) < 0)
return AVERROR(ENOMEM);
avio_put_str16le(dyn_buf, desc);
len = url_close_dyn_buf(dyn_buf, &buf);
put_le16(pb, len / 2); // "number of characters" = length in bytes / 2
put_buffer(pb, buf, len);
av_freep(&buf);
put_le16(pb, 0); /* no parameters */
/* id */
if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {
put_le16(pb, 2);
put_le16(pb, enc->codec_tag);
} else {
put_le16(pb, 4);
put_le32(pb, enc->codec_tag);
}
if(!enc->codec_tag)
return -1;
}
end_header(pb, hpos);
/* patch the header size fields */
cur_pos = url_ftell(pb);
header_size = cur_pos - header_offset;
if (asf->is_streamed) {
header_size += 8 + 30 + 50;
url_fseek(pb, header_offset - 10 - 30, SEEK_SET);
put_le16(pb, header_size);
url_fseek(pb, header_offset - 2 - 30, SEEK_SET);
put_le16(pb, header_size);
header_size -= 8 + 30 + 50;
}
header_size += 24 + 6;
url_fseek(pb, header_offset - 14, SEEK_SET);
put_le64(pb, header_size);
url_fseek(pb, cur_pos, SEEK_SET);
/* movie chunk, followed by packets of packet_size */
asf->data_offset = cur_pos;
put_guid(pb, &ff_asf_data_header);
put_le64(pb, data_chunk_size);
put_guid(pb, &ff_asf_my_guid);
put_le64(pb, asf->nb_packets); /* nb packets */
put_byte(pb, 1); /* ??? */
put_byte(pb, 1); /* ??? */
return 0;
}
int ff_mov_init_hinting(AVFormatContext *s, int index, int src_index)
{
MOVMuxContext *mov = s->priv_data;
MOVTrack *track = &mov->tracks[index];
MOVTrack *src_track = &mov->tracks[src_index];
AVStream *src_st = s->streams[src_index];
int ret = AVERROR(ENOMEM);
AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
track->tag = MKTAG('r','t','p',' ');
track->src_track = src_index;
if (!rtp_format) {
ret = AVERROR(ENOENT);
goto fail;
}
track->enc = avcodec_alloc_context();
if (!track->enc)
goto fail;
track->enc->codec_type = AVMEDIA_TYPE_DATA;
track->enc->codec_tag = track->tag;
track->rtp_ctx = avformat_alloc_context();
if (!track->rtp_ctx)
goto fail;
track->rtp_ctx->oformat = rtp_format;
if (!av_new_stream(track->rtp_ctx, 0))
goto fail;
/* Copy stream parameters */
track->rtp_ctx->streams[0]->sample_aspect_ratio =
src_st->sample_aspect_ratio;
/* Remove the allocated codec context, link to the original one
* instead, to give the rtp muxer access to codec parameters. */
av_free(track->rtp_ctx->streams[0]->codec);
track->rtp_ctx->streams[0]->codec = src_st->codec;
if ((ret = url_open_dyn_packet_buf(&track->rtp_ctx->pb,
RTP_MAX_PACKET_SIZE)) < 0)
goto fail;
ret = av_write_header(track->rtp_ctx);
if (ret)
goto fail;
/* Copy the RTP AVStream timebase back to the hint AVStream */
track->timescale = track->rtp_ctx->streams[0]->time_base.den;
/* Mark the hinted track that packets written to it should be
* sent to this track for hinting. */
src_track->hint_track = index;
return 0;
fail:
av_log(s, AV_LOG_WARNING,
"Unable to initialize hinting of stream %d\n", src_index);
if (track->rtp_ctx && track->rtp_ctx->pb) {
uint8_t *buf;
url_close_dyn_buf(track->rtp_ctx->pb, &buf);
av_free(buf);
}
if (track->rtp_ctx && track->rtp_ctx->streams[0]) {
av_metadata_free(&track->rtp_ctx->streams[0]->metadata);
av_free(track->rtp_ctx->streams[0]);
}
if (track->rtp_ctx) {
av_metadata_free(&track->rtp_ctx->metadata);
av_free(track->rtp_ctx->priv_data);
av_freep(&track->rtp_ctx);
}
av_freep(&track->enc);
/* Set a default timescale, to avoid crashes in dump_format */
track->timescale = 90000;
return ret;
}