uint32_t client_report_size; ///< number of bytes after which client should report to server
uint32_t bytes_read; ///< number of bytes read from server
uint32_t last_bytes_read; ///< number of bytes read last reported to server
+ uint32_t last_timestamp; ///< last timestamp received in a packet
int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
int has_audio; ///< presence of audio data
int has_video; ///< presence of video data
*value = '\0';
value++;
- if (!field || !value)
- goto fail;
-
ff_amf_write_field_name(p, field);
} else {
goto fail;
return rtmp_send_packet(rt, &pkt, 1);
}
+/**
+ * Generate a pause packet that either pauses or unpauses the current stream.
+ */
+static int gen_pause(URLContext *s, RTMPContext *rt, int pause, uint32_t timestamp)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+ int ret;
+
+ av_log(s, AV_LOG_DEBUG, "Sending pause command for timestamp %d\n",
+ timestamp);
+
+ if ((ret = ff_rtmp_packet_create(&pkt, 3, RTMP_PT_INVOKE, 0, 29)) < 0)
+ return ret;
+
+ pkt.extra = rt->stream_id;
+
+ p = pkt.data;
+ ff_amf_write_string(&p, "pause");
+ ff_amf_write_number(&p, 0); //no tracking back responses
+ ff_amf_write_null(&p); //as usual, the first null param
+ ff_amf_write_bool(&p, pause); // pause or unpause
+ ff_amf_write_number(&p, timestamp); //where we pause the stream
+
+ return rtmp_send_packet(rt, &pkt, 1);
+}
+
/**
* Generate 'publish' call and send it to the server.
*/
/* Gracefully ignore Adobe-specific historical artifact errors. */
level = AV_LOG_WARNING;
ret = 0;
+ } else if (tracked_method && !strcmp(tracked_method, "getStreamLength")) {
+ level = rt->live ? AV_LOG_DEBUG : AV_LOG_WARNING;
+ ret = 0;
} else if (tracked_method && !strcmp(tracked_method, "connect")) {
ret = handle_connect_error(s, tmpstr);
if (!ret) {
switch (pkt->type) {
case RTMP_PT_BYTES_READ:
- av_dlog(s, "received bytes read report\n");
+ av_log(s, AV_LOG_TRACE, "received bytes read report\n");
break;
case RTMP_PT_CHUNK_SIZE:
if ((ret = handle_chunk_size(s, pkt)) < 0)
return AVERROR(EIO);
}
}
+
+ // Track timestamp for later use
+ rt->last_timestamp = rpkt.timestamp;
+
rt->bytes_read += ret;
if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
}
if (rt->is_input) {
- int err;
// generate FLV header for demuxer
rt->flv_size = 13;
- if ((err = av_reallocp(&rt->flv_data, rt->flv_size)) < 0)
- return err;
+ if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0)
+ goto fail;
rt->flv_off = 0;
memcpy(rt->flv_data, "FLV\1\0\0\0\0\011\0\0\0\0", rt->flv_size);
// audio or video packet arrives.
while (!rt->has_audio && !rt->has_video && !rt->received_metadata) {
if ((ret = get_packet(s, 0)) < 0)
- return ret;
+ goto fail;
}
// Either after we have read the metadata or (if there is none) the
return timestamp;
}
+static int rtmp_pause(URLContext *s, int pause)
+{
+ RTMPContext *rt = s->priv_data;
+ int ret;
+ av_log(s, AV_LOG_DEBUG, "Pause at timestamp %d\n",
+ rt->last_timestamp);
+ if ((ret = gen_pause(s, rt, pause, rt->last_timestamp)) < 0) {
+ av_log(s, AV_LOG_ERROR, "Unable to send pause command at timestamp %d\n",
+ rt->last_timestamp);
+ return ret;
+ }
+ return 0;
+}
+
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
{
RTMPContext *rt = s->priv_data;
int size_temp = size;
- int pktsize, pkttype;
+ int pktsize, pkttype, copy;
uint32_t ts;
const uint8_t *buf_temp = buf;
uint8_t c;
if (rt->flv_header_bytes < RTMP_HEADER) {
const uint8_t *header = rt->flv_header;
- int copy = FFMIN(RTMP_HEADER - rt->flv_header_bytes, size_temp);
int channel = RTMP_AUDIO_CHANNEL;
+ copy = FFMIN(RTMP_HEADER - rt->flv_header_bytes, size_temp);
bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
rt->flv_header_bytes += copy;
size_temp -= copy;
if (pkttype == RTMP_PT_VIDEO)
channel = RTMP_VIDEO_CHANNEL;
- //force 12bytes header
if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
pkttype == RTMP_PT_NOTIFY) {
- if (pkttype == RTMP_PT_NOTIFY)
- pktsize += 16;
if ((ret = ff_rtmp_check_alloc_array(&rt->prev_pkt[1],
&rt->nb_prev_pkt[1],
channel)) < 0)
return ret;
+ // Force sending a full 12 bytes header by clearing the
+ // channel id, to make it not match a potential earlier
+ // packet in the same channel.
rt->prev_pkt[1][channel].channel_id = 0;
}
rt->out_pkt.extra = rt->stream_id;
rt->flv_data = rt->out_pkt.data;
-
- if (pkttype == RTMP_PT_NOTIFY)
- ff_amf_write_string(&rt->flv_data, "@setDataFrame");
}
- if (rt->flv_size - rt->flv_off > size_temp) {
- bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
- rt->flv_off += size_temp;
- size_temp = 0;
- } else {
- bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
- size_temp -= rt->flv_size - rt->flv_off;
- rt->flv_off += rt->flv_size - rt->flv_off;
- }
+ copy = FFMIN(rt->flv_size - rt->flv_off, size_temp);
+ bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, copy);
+ rt->flv_off += copy;
+ size_temp -= copy;
if (rt->flv_off == rt->flv_size) {
rt->skip_bytes = 4;
+ if (rt->out_pkt.type == RTMP_PT_NOTIFY) {
+ // For onMetaData and |RtmpSampleAccess packets, we want
+ // @setDataFrame prepended to the packet before it gets sent.
+ // However, not all RTMP_PT_NOTIFY packets (e.g., onTextData
+ // and onCuePoint).
+ uint8_t commandbuffer[64];
+ int stringlen = 0;
+ GetByteContext gbc;
+
+ bytestream2_init(&gbc, rt->flv_data, rt->flv_size);
+ if (!ff_amf_read_string(&gbc, commandbuffer, sizeof(commandbuffer),
+ &stringlen)) {
+ if (!strcmp(commandbuffer, "onMetaData") ||
+ !strcmp(commandbuffer, "|RtmpSampleAccess")) {
+ uint8_t *ptr;
+ if ((ret = av_reallocp(&rt->out_pkt.data, rt->out_pkt.size + 16)) < 0) {
+ rt->flv_size = rt->flv_off = rt->flv_header_bytes = 0;
+ return ret;
+ }
+ memmove(rt->out_pkt.data + 16, rt->out_pkt.data, rt->out_pkt.size);
+ rt->out_pkt.size += 16;
+ ptr = rt->out_pkt.data;
+ ff_amf_write_string(&ptr, "@setDataFrame");
+ }
+ }
+ }
+
if ((ret = rtmp_send_packet(rt, &rt->out_pkt, 0)) < 0)
return ret;
rt->flv_size = 0;
.url_open = rtmp_open, \
.url_read = rtmp_read, \
.url_read_seek = rtmp_seek, \
+ .url_read_pause = rtmp_pause, \
.url_write = rtmp_write, \
.url_close = rtmp_close, \
.priv_data_size = sizeof(RTMPContext), \
projects / ffmpeg / blobdiff