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authorMichael Niedermayer <michaelni@gmx.at>2012-06-18 20:05:32 +0200
committerMichael Niedermayer <michaelni@gmx.at>2012-06-18 20:07:00 +0200
commit82edf6727f0663601351081ca1e4fb20d1752972 (patch)
tree12479c3ec8cedfa0ec4dda38a72023224f2b5b73 /libavformat/rtmphttp.c
parentf87dacb27de93f995cb18f9dcc73581ef8fc157b (diff)
parentf61ce90caa909d131ea6ec205823568a38115529 (diff)
downloadffmpeg-82edf6727f0663601351081ca1e4fb20d1752972.tar.gz
Merge remote-tracking branch 'qatar/master'
* qatar/master: lavr: add x86-optimized functions for mixing 1-to-2 s16p with flt coeffs lavr: add x86-optimized functions for mixing 1-to-2 fltp with flt coeffs Add Dolby/DPLII downmix support to libavresample vorbisdec: replace div/mod in loop with a counter fate: vorbis: add 5.1 surround test rtpenc: Allow requesting H264 RTP packetization mode 0 configure: Sort the library listings in the help text alphabetically dwt: remove variable-length arrays RTMPT protocol support http: Properly handle chunked transfer-encoding for replies to post data http: Fail reading if the connection has gone away amr: Mark an array const amr: More space cleanup rtpenc: Fix memory leaks in the muxer open function Conflicts: Changelog configure doc/APIchanges libavformat/version.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/rtmphttp.c')
-rw-r--r--libavformat/rtmphttp.c239
1 files changed, 239 insertions, 0 deletions
diff --git a/libavformat/rtmphttp.c b/libavformat/rtmphttp.c
new file mode 100644
index 0000000000..c431d28853
--- /dev/null
+++ b/libavformat/rtmphttp.c
@@ -0,0 +1,239 @@
+/*
+ * RTMP HTTP network protocol
+ * Copyright (c) 2012 Samuel Pitoiset
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * RTMP HTTP protocol
+ */
+
+#include "libavutil/avstring.h"
+#include "libavutil/intfloat.h"
+#include "libavutil/opt.h"
+#include "internal.h"
+#include "http.h"
+
+#define RTMPT_DEFAULT_PORT 80
+
+/* protocol handler context */
+typedef struct RTMP_HTTPContext {
+ URLContext *stream; ///< HTTP stream
+ char host[256]; ///< hostname of the server
+ int port; ///< port to connect (default is 80)
+ char client_id[64]; ///< client ID used for all requests except the first one
+ int seq; ///< sequence ID used for all requests
+ uint8_t *out_data; ///< output buffer
+ int out_size; ///< current output buffer size
+ int out_capacity; ///< current output buffer capacity
+ int initialized; ///< flag indicating when the http context is initialized
+ int finishing; ///< flag indicating when the client closes the connection
+} RTMP_HTTPContext;
+
+static int rtmp_http_send_cmd(URLContext *h, const char *cmd)
+{
+ RTMP_HTTPContext *rt = h->priv_data;
+ char uri[2048];
+ uint8_t c;
+ int ret;
+
+ ff_url_join(uri, sizeof(uri), "http", NULL, rt->host, rt->port,
+ "/%s/%s/%d", cmd, rt->client_id, rt->seq++);
+
+ av_opt_set_bin(rt->stream->priv_data, "post_data", rt->out_data,
+ rt->out_size, 0);
+
+ /* send a new request to the server */
+ if ((ret = ff_http_do_new_request(rt->stream, uri)) < 0)
+ return ret;
+
+ /* re-init output buffer */
+ rt->out_size = 0;
+
+ /* read the first byte which contains the polling interval */
+ if ((ret = ffurl_read(rt->stream, &c, 1)) < 0)
+ return ret;
+
+ return ret;
+}
+
+static int rtmp_http_write(URLContext *h, const uint8_t *buf, int size)
+{
+ RTMP_HTTPContext *rt = h->priv_data;
+ void *ptr;
+
+ if (rt->out_size + size > rt->out_capacity) {
+ rt->out_capacity = (rt->out_size + size) * 2;
+ ptr = av_realloc(rt->out_data, rt->out_capacity);
+ if (!ptr)
+ return AVERROR(ENOMEM);
+ rt->out_data = ptr;
+ }
+
+ memcpy(rt->out_data + rt->out_size, buf, size);
+ rt->out_size += size;
+
+ return size;
+}
+
+static int rtmp_http_read(URLContext *h, uint8_t *buf, int size)
+{
+ RTMP_HTTPContext *rt = h->priv_data;
+ int ret, off = 0;
+
+ /* try to read at least 1 byte of data */
+ do {
+ ret = ffurl_read(rt->stream, buf + off, size);
+ if (ret < 0 && ret != AVERROR_EOF)
+ return ret;
+
+ if (ret == AVERROR_EOF) {
+ if (rt->finishing) {
+ /* Do not send new requests when the client wants to
+ * close the connection. */
+ return AVERROR(EAGAIN);
+ }
+
+ /* When the client has reached end of file for the last request,
+ * we have to send a new request if we have buffered data.
+ * Otherwise, we have to send an idle POST. */
+ if (rt->out_size > 0) {
+ if ((ret = rtmp_http_send_cmd(h, "send")) < 0)
+ return ret;
+ } else {
+ if ((ret = rtmp_http_write(h, "", 1)) < 0)
+ return ret;
+
+ if ((ret = rtmp_http_send_cmd(h, "idle")) < 0)
+ return ret;
+ }
+
+ if (h->flags & AVIO_FLAG_NONBLOCK) {
+ /* no incoming data to handle in nonblocking mode */
+ return AVERROR(EAGAIN);
+ }
+ } else {
+ off += ret;
+ size -= ret;
+ }
+ } while (off <= 0);
+
+ return off;
+}
+
+static int rtmp_http_close(URLContext *h)
+{
+ RTMP_HTTPContext *rt = h->priv_data;
+ uint8_t tmp_buf[2048];
+ int ret = 0;
+
+ if (rt->initialized) {
+ /* client wants to close the connection */
+ rt->finishing = 1;
+
+ do {
+ ret = rtmp_http_read(h, tmp_buf, sizeof(tmp_buf));
+ } while (ret > 0);
+
+ /* re-init output buffer before sending the close command */
+ rt->out_size = 0;
+
+ if ((ret = rtmp_http_write(h, "", 1)) == 1)
+ ret = rtmp_http_send_cmd(h, "close");
+ }
+
+ av_freep(&rt->out_data);
+ ffurl_close(rt->stream);
+
+ return ret;
+}
+
+static int rtmp_http_open(URLContext *h, const char *uri, int flags)
+{
+ RTMP_HTTPContext *rt = h->priv_data;
+ char headers[1024], url[1024];
+ int ret, off = 0;
+
+ av_url_split(NULL, 0, NULL, 0, rt->host, sizeof(rt->host), &rt->port,
+ NULL, 0, uri);
+
+ if (rt->port < 0)
+ rt->port = RTMPT_DEFAULT_PORT;
+
+ /* This is the first request that is sent to the server in order to
+ * register a client on the server and start a new session. The server
+ * replies with a unique id (usually a number) that is used by the client
+ * for all future requests.
+ * Note: the reply doesn't contain a value for the polling interval.
+ * A successful connect resets the consecutive index that is used
+ * in the URLs. */
+ ff_url_join(url, sizeof(url), "http", NULL, rt->host, rt->port, "/open/1");
+
+ /* alloc the http context */
+ if ((ret = ffurl_alloc(&rt->stream, url, AVIO_FLAG_READ_WRITE, NULL)) < 0)
+ goto fail;
+
+ /* set options */
+ snprintf(headers, sizeof(headers),
+ "Cache-Control: no-cache\r\n"
+ "Content-type: application/x-fcs\r\n"
+ "User-Agent: Shockwave Flash\r\n");
+ av_opt_set(rt->stream->priv_data, "headers", headers, 0);
+ av_opt_set(rt->stream->priv_data, "multiple_requests", "1", 0);
+ av_opt_set_bin(rt->stream->priv_data, "post_data", "", 1, 0);
+
+ /* open the http context */
+ if ((ret = ffurl_connect(rt->stream, NULL)) < 0)
+ goto fail;
+
+ /* read the server reply which contains a unique ID */
+ for (;;) {
+ ret = ffurl_read(rt->stream, rt->client_id + off, sizeof(rt->client_id) - off);
+ if (ret == AVERROR_EOF)
+ break;
+ if (ret < 0)
+ goto fail;
+ off += ret;
+ if (off == sizeof(rt->client_id)) {
+ ret = AVERROR(EIO);
+ goto fail;
+ }
+ }
+ while (off > 0 && isspace(rt->client_id[off - 1]))
+ off--;
+ rt->client_id[off] = '\0';
+
+ /* http context is now initialized */
+ rt->initialized = 1;
+ return 0;
+
+fail:
+ rtmp_http_close(h);
+ return ret;
+}
+
+URLProtocol ff_rtmphttp_protocol = {
+ .name = "rtmphttp",
+ .url_open = rtmp_http_open,
+ .url_read = rtmp_http_read,
+ .url_write = rtmp_http_write,
+ .url_close = rtmp_http_close,
+ .priv_data_size = sizeof(RTMP_HTTPContext),
+ .flags = URL_PROTOCOL_FLAG_NETWORK,
+};