]> git.sesse.net Git - ffmpeg/blobdiff - libavformat/rtmpproto.c
mov: Fix little endian audio detection
[ffmpeg] / libavformat / rtmpproto.c
index 8d8aabc60a53481a439ffc00cb3d7fbe904e28da..be87b4d4c15144032a4d8754500533a4b48d2fa1 100644 (file)
@@ -96,7 +96,11 @@ typedef struct RTMPContext {
     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
+    int           received_metadata;          ///< Indicates if we have received metadata about the streams
     uint8_t       flv_header[RTMP_HEADER];    ///< partial incoming flv packet header
     int           flv_header_bytes;           ///< number of initialized bytes in flv_header
     int           nb_invokes;                 ///< keeps track of invoke messages
@@ -120,6 +124,7 @@ typedef struct RTMPContext {
     int           listen;                     ///< listen mode flag
     int           listen_timeout;             ///< listen timeout to wait for new connections
     int           nb_streamid;                ///< The next stream id to return on createStream calls
+    double        duration;                   ///< Duration of the stream in seconds as returned by the server (only valid if non-zero)
     char          username[50];
     char          password[50];
     char          auth_params[500];
@@ -271,9 +276,6 @@ static int rtmp_write_amf_data(URLContext *s, char *param, uint8_t **p)
         *value = '\0';
         value++;
 
-        if (!field || !value)
-            goto fail;
-
         ff_amf_write_field_name(p, field);
     } else {
         goto fail;
@@ -372,7 +374,7 @@ static int gen_connect(URLContext *s, RTMPContext *rt)
         char *param = rt->conn;
 
         // Write arbitrary AMF data to the Connect message.
-        while (param != NULL) {
+        while (param) {
             char *sep;
             param += strspn(param, " ");
             if (!*param)
@@ -502,7 +504,7 @@ static int read_connect(URLContext *s, RTMPContext *rt)
     if (ret < 0)
         return ret;
 
-    // Send result_ NetConnection.Connect.Success to connect
+    // Send _result NetConnection.Connect.Success to connect
     if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL,
                                      RTMP_PT_INVOKE, 0,
                                      RTMP_PKTDATA_DEFAULT_SIZE)) < 0)
@@ -675,6 +677,30 @@ static int gen_delete_stream(URLContext *s, RTMPContext *rt)
     return rtmp_send_packet(rt, &pkt, 0);
 }
 
+/**
+ * Generate 'getStreamLength' call and send it to the server. If the server
+ * knows the duration of the selected stream, it will reply with the duration
+ * in seconds.
+ */
+static int gen_get_stream_length(URLContext *s, RTMPContext *rt)
+{
+    RTMPPacket pkt;
+    uint8_t *p;
+    int ret;
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 31 + strlen(rt->playpath))) < 0)
+        return ret;
+
+    p = pkt.data;
+    ff_amf_write_string(&p, "getStreamLength");
+    ff_amf_write_number(&p, ++rt->nb_invokes);
+    ff_amf_write_null(&p);
+    ff_amf_write_string(&p, rt->playpath);
+
+    return rtmp_send_packet(rt, &pkt, 1);
+}
+
 /**
  * Generate client buffer time and send it to the server.
  */
@@ -747,6 +773,33 @@ static int gen_seek(URLContext *s, RTMPContext *rt, int64_t timestamp)
     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.
  */
@@ -1763,6 +1816,9 @@ static int handle_invoke_error(URLContext *s, RTMPPacket *pkt)
             /* 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) {
@@ -1950,6 +2006,45 @@ static int send_invoke_response(URLContext *s, RTMPPacket *pkt)
     return ret;
 }
 
+/**
+ * Read the AMF_NUMBER response ("_result") to a function call
+ * (e.g. createStream()). This response should be made up of the AMF_STRING
+ * "result", a NULL object and then the response encoded as AMF_NUMBER. On a
+ * successful response, we will return set the value to number (otherwise number
+ * will not be changed).
+ *
+ * @return 0 if reading the value succeeds, negative value otherwiss
+ */
+static int read_number_result(RTMPPacket *pkt, double *number)
+{
+    // We only need to fit "_result" in this.
+    uint8_t strbuffer[8];
+    int stringlen;
+    double numbuffer;
+    GetByteContext gbc;
+
+    bytestream2_init(&gbc, pkt->data, pkt->size);
+
+    // Value 1/4: "_result" as AMF_STRING
+    if (ff_amf_read_string(&gbc, strbuffer, sizeof(strbuffer), &stringlen))
+        return AVERROR_INVALIDDATA;
+    if (strcmp(strbuffer, "_result"))
+        return AVERROR_INVALIDDATA;
+    // Value 2/4: The callee reference number
+    if (ff_amf_read_number(&gbc, &numbuffer))
+        return AVERROR_INVALIDDATA;
+    // Value 3/4: Null
+    if (ff_amf_read_null(&gbc))
+        return AVERROR_INVALIDDATA;
+    // Value 4/4: The resonse as AMF_NUMBER
+    if (ff_amf_read_number(&gbc, &numbuffer))
+        return AVERROR_INVALIDDATA;
+    else
+        *number = numbuffer;
+
+    return 0;
+}
+
 static int handle_invoke_result(URLContext *s, RTMPPacket *pkt)
 {
     RTMPContext *rt = s->priv_data;
@@ -1991,22 +2086,30 @@ static int handle_invoke_result(URLContext *s, RTMPPacket *pkt)
             }
         }
     } else if (!strcmp(tracked_method, "createStream")) {
-        //extract a number from the result
-        if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
+        double stream_id;
+        if (read_number_result(pkt, &stream_id)) {
             av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
         } else {
-            rt->stream_id = av_int2double(AV_RB64(pkt->data + 21));
+            rt->stream_id = stream_id;
         }
 
         if (!rt->is_input) {
             if ((ret = gen_publish(s, rt)) < 0)
                 goto fail;
         } else {
+            if (rt->live != -1) {
+                if ((ret = gen_get_stream_length(s, rt)) < 0)
+                    goto fail;
+            }
             if ((ret = gen_play(s, rt)) < 0)
                 goto fail;
             if ((ret = gen_buffer_time(s, rt)) < 0)
                 goto fail;
         }
+    } else if (!strcmp(tracked_method, "getStreamLength")) {
+        if (read_number_result(pkt, &rt->duration)) {
+            av_log(s, AV_LOG_WARNING, "Unexpected reply on getStreamLength()\n");
+        }
     }
 
 fail:
@@ -2109,6 +2212,12 @@ static int append_flv_data(RTMPContext *rt, RTMPPacket *pkt, int skip)
     const int size      = pkt->size - skip;
     uint32_t ts         = pkt->timestamp;
 
+    if (pkt->type == RTMP_PT_AUDIO) {
+        rt->has_audio = 1;
+    } else if (pkt->type == RTMP_PT_VIDEO) {
+        rt->has_video = 1;
+    }
+
     old_flv_size = update_offset(rt, size + 15);
 
     if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0) {
@@ -2141,6 +2250,38 @@ static int handle_notify(URLContext *s, RTMPPacket *pkt)
                            &stringlen))
         return AVERROR_INVALIDDATA;
 
+    if (!strcmp(commandbuffer, "onMetaData")) {
+        // metadata properties should be stored in a mixed array
+        if (bytestream2_get_byte(&gbc) == AMF_DATA_TYPE_MIXEDARRAY) {
+            // We have found a metaData Array so flv can determine the streams
+            // from this.
+            rt->received_metadata = 1;
+            // skip 32-bit max array index
+            bytestream2_skip(&gbc, 4);
+            while (bytestream2_get_bytes_left(&gbc) > 3) {
+                if (ff_amf_get_string(&gbc, statusmsg, sizeof(statusmsg),
+                                      &stringlen))
+                    return AVERROR_INVALIDDATA;
+                // We do not care about the content of the property (yet).
+                stringlen = ff_amf_tag_size(gbc.buffer, gbc.buffer_end);
+                if (stringlen < 0)
+                    return AVERROR_INVALIDDATA;
+                bytestream2_skip(&gbc, stringlen);
+
+                // The presence of the following properties indicates that the
+                // respective streams are present.
+                if (!strcmp(statusmsg, "videocodecid")) {
+                    rt->has_video = 1;
+                }
+                if (!strcmp(statusmsg, "audiocodecid")) {
+                    rt->has_audio = 1;
+                }
+            }
+            if (bytestream2_get_be24(&gbc) != AMF_END_OF_OBJECT)
+                return AVERROR_INVALIDDATA;
+        }
+    }
+
     // Skip the @setDataFrame string and validate it is a notification
     if (!strcmp(commandbuffer, "@setDataFrame")) {
         skip = gbc.buffer - pkt->data;
@@ -2283,6 +2424,10 @@ static int get_packet(URLContext *s, int for_header)
                 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");
@@ -2369,6 +2514,70 @@ static int rtmp_close(URLContext *h)
     return ret;
 }
 
+/**
+ * Insert a fake onMetadata packet into the FLV stream to notify the FLV
+ * demuxer about the duration of the stream.
+ *
+ * This should only be done if there was no real onMetadata packet sent by the
+ * server at the start of the stream and if we were able to retrieve a valid
+ * duration via a getStreamLength call.
+ *
+ * @return 0 for successful operation, negative value in case of error
+ */
+static int inject_fake_duration_metadata(RTMPContext *rt)
+{
+    // We need to insert the metdata packet directly after the FLV
+    // header, i.e. we need to move all other already read data by the
+    // size of our fake metadata packet.
+
+    uint8_t* p;
+    // Keep old flv_data pointer
+    uint8_t* old_flv_data = rt->flv_data;
+    // Allocate a new flv_data pointer with enough space for the additional package
+    if (!(rt->flv_data = av_malloc(rt->flv_size + 55))) {
+        rt->flv_data = old_flv_data;
+        return AVERROR(ENOMEM);
+    }
+
+    // Copy FLV header
+    memcpy(rt->flv_data, old_flv_data, 13);
+    // Copy remaining packets
+    memcpy(rt->flv_data + 13 + 55, old_flv_data + 13, rt->flv_size - 13);
+    // Increase the size by the injected packet
+    rt->flv_size += 55;
+    // Delete the old FLV data
+    av_free(old_flv_data);
+
+    p = rt->flv_data + 13;
+    bytestream_put_byte(&p, FLV_TAG_TYPE_META);
+    bytestream_put_be24(&p, 40); // size of data part (sum of all parts below)
+    bytestream_put_be24(&p, 0);  // timestamp
+    bytestream_put_be32(&p, 0);  // reserved
+
+    // first event name as a string
+    bytestream_put_byte(&p, AMF_DATA_TYPE_STRING);
+    // "onMetaData" as AMF string
+    bytestream_put_be16(&p, 10);
+    bytestream_put_buffer(&p, "onMetaData", 10);
+
+    // mixed array (hash) with size and string/type/data tuples
+    bytestream_put_byte(&p, AMF_DATA_TYPE_MIXEDARRAY);
+    bytestream_put_be32(&p, 1); // metadata_count
+
+    // "duration" as AMF string
+    bytestream_put_be16(&p, 8);
+    bytestream_put_buffer(&p, "duration", 8);
+    bytestream_put_byte(&p, AMF_DATA_TYPE_NUMBER);
+    bytestream_put_be64(&p, av_double2int(rt->duration));
+
+    // Finalise object
+    bytestream_put_be16(&p, 0); // Empty string
+    bytestream_put_byte(&p, AMF_END_OF_OBJECT);
+    bytestream_put_be32(&p, 40); // size of data part (sum of all parts below)
+
+    return 0;
+}
+
 /**
  * Open RTMP connection and verify that the stream can be played.
  *
@@ -2382,7 +2591,7 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
 {
     RTMPContext *rt = s->priv_data;
     char proto[8], hostname[256], path[1024], auth[100], *fname;
-    char *old_app;
+    char *old_app, *qmark, fname_buffer[1024];
     uint8_t buf[2048];
     int port;
     AVDictionary *opts = NULL;
@@ -2480,7 +2689,20 @@ reconnect:
     }
 
     //extract "app" part from path
-    if (!strncmp(path, "/ondemand/", 10)) {
+    qmark = strchr(path, '?');
+    if (qmark && strstr(qmark, "slist=")) {
+        char* amp;
+        // After slist we have the playpath, before the params, the app
+        av_strlcpy(rt->app, path + 1, FFMIN(qmark - path, APP_MAX_LENGTH));
+        fname = strstr(path, "slist=") + 6;
+        // Strip any further query parameters from fname
+        amp = strchr(fname, '&');
+        if (amp) {
+            av_strlcpy(fname_buffer, fname, FFMIN(amp - fname + 1,
+                                                  sizeof(fname_buffer)));
+            fname = fname_buffer;
+        }
+    } else if (!strncmp(path, "/ondemand/", 10)) {
         fname = path + 10;
         memcpy(rt->app, "ondemand", 9);
     } else {
@@ -2495,10 +2717,10 @@ reconnect:
             fname = strchr(p + 1, '/');
             if (!fname || (c && c < fname)) {
                 fname = p + 1;
-                av_strlcpy(rt->app, path + 1, p - path);
+                av_strlcpy(rt->app, path + 1, FFMIN(p - path, APP_MAX_LENGTH));
             } else {
                 fname++;
-                av_strlcpy(rt->app, path + 1, fname - path - 1);
+                av_strlcpy(rt->app, path + 1, FFMIN(fname - path - 1, APP_MAX_LENGTH));
             }
         }
     }
@@ -2558,8 +2780,12 @@ reconnect:
 
     rt->client_report_size = 1048576;
     rt->bytes_read = 0;
+    rt->has_audio = 0;
+    rt->has_video = 0;
+    rt->received_metadata = 0;
     rt->last_bytes_read = 0;
     rt->server_bw = 2500000;
+    rt->duration = 0;
 
     av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
            proto, path, rt->app, rt->playpath);
@@ -2591,13 +2817,40 @@ reconnect:
     }
 
     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\5\0\0\0\011\0\0\0\0", rt->flv_size);
+        memcpy(rt->flv_data, "FLV\1\0\0\0\0\011\0\0\0\0", rt->flv_size);
+
+        // Read packets until we reach the first A/V packet or read metadata.
+        // If there was a metadata package in front of the A/V packets, we can
+        // build the FLV header from this. If we do not receive any metadata,
+        // the FLV decoder will allocate the needed streams when their first
+        // audio or video packet arrives.
+        while (!rt->has_audio && !rt->has_video && !rt->received_metadata) {
+            if ((ret = get_packet(s, 0)) < 0)
+               goto fail;
+        }
+
+        // Either after we have read the metadata or (if there is none) the
+        // first packet of an A/V stream, we have a better knowledge about the
+        // streams, so set the FLV header accordingly.
+        if (rt->has_audio) {
+            rt->flv_data[4] |= FLV_HEADER_FLAG_HASAUDIO;
+        }
+        if (rt->has_video) {
+            rt->flv_data[4] |= FLV_HEADER_FLAG_HASVIDEO;
+        }
+
+        // If we received the first packet of an A/V stream and no metadata but
+        // the server returned a valid duration, create a fake metadata packet
+        // to inform the FLV decoder about the duration.
+        if (!rt->received_metadata && rt->duration > 0) {
+            if ((ret = inject_fake_duration_metadata(rt)) < 0)
+                goto fail;
+        }
     } else {
         rt->flv_size = 0;
         rt->flv_data = NULL;
@@ -2662,11 +2915,25 @@ static int64_t rtmp_seek(URLContext *s, int stream_index, int64_t timestamp,
     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;
@@ -2683,8 +2950,8 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
 
         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;
@@ -2701,15 +2968,15 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
             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;
             }
 
@@ -2720,24 +2987,43 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
 
             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;
@@ -2824,6 +3110,7 @@ URLProtocol ff_##flavor##_protocol = {           \
     .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),       \