Voicemail: get correct duration when copying file to vm

Changes made during format improvements resulted in the
recording to voicemail option 'm' of the MixMonitor app
writing a zero length duration in the msgXXXX.txt file.

This change introduces a new function ast_ratestream(),
which provides the sample rate of the format associated
with the stream, and updates the app_voicemail function
for ast_app_copy_recording_to_vm to calculate the right
duration.

Review: https://reviewboard.asterisk.org/r/3996/
ASTERISK-24328 #close
........

Merged revisions 423192 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index fdfbf225514a280023ae54d445013d671e7c5708..45f9ebfea5316a77d2e42c517f47d9f69354fc12 100644 (file)
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -6130,15 +6130,11 @@ static int msg_create_from_file(struct ast_vm_recording_data *recdata)
        if ((recording_fs = ast_readfile(recdata->recording_file, recdata->recording_ext, NULL, 0, 0, VOICEMAIL_DIR_MODE))) {
                if (!ast_seekstream(recording_fs, 0, SEEK_END)) {
                        long framelength = ast_tellstream(recording_fs);
-                       struct ast_format *result;
-                       /* XXX This use of ast_getformatbyname seems incorrect here. The file extension does not necessarily correspond
-                        * to the name of the format. For instance, if "raw" were passed in, I don't think ast_getformatbyname would
-                        * find the slinear format
-                        */
-                       result = ast_format_cache_get(recdata->recording_ext);
-                       if (result) {
-                               duration = (int) (framelength / ast_format_get_sample_rate(result));
-                               ao2_ref(result, -1);
+                       int sample_rate = ast_ratestream(recording_fs);
+                       if (sample_rate) {
+                               duration = (int) (framelength / sample_rate);
+                       } else {
+                               ast_log(LOG_ERROR,"Unable to determine sample rate of recording %s\n", recdata->recording_file);
                        }
                }
        }

diff --git a/include/asterisk/file.h b/include/asterisk/file.h
index 3d8d2c97c33362690781629979daba970995350f..c71866ead54f7a2d082cc3ec0546e6bbb36bcd05 100644 (file)
--- a/include/asterisk/file.h
+++ b/include/asterisk/file.h
@@ -350,6 +350,13 @@ int ast_stream_rewind(struct ast_filestream *fs, off_t ms);
  */
 off_t ast_tellstream(struct ast_filestream *fs);
 
+/*!
+ * \brief Return the sample rate of the stream's format
+ * \param fs fs to act on
+ * \return sample rate in Hz
+ */
+int ast_ratestream(struct ast_filestream *fs);
+
 /*! 
  * \brief Read a frame from a filestream 
  * \param s ast_filestream to act on

diff --git a/main/file.c b/main/file.c
index 393832cdd31735c8ed4f9255b54d98fd088e5377..1d3755ac28e43618707e65d1d8c370274938aca7 100644 (file)
--- a/main/file.c
+++ b/main/file.c
@@ -1026,6 +1026,11 @@ off_t ast_tellstream(struct ast_filestream *fs)
        return fs->fmt->tell(fs);
 }
 
+int ast_ratestream(struct ast_filestream *fs)
+{
+       return ast_format_get_sample_rate(fs->fmt->format);
+}
+
 int ast_stream_fastforward(struct ast_filestream *fs, off_t ms)
 {
        return ast_seekstream(fs, ms * DEFAULT_SAMPLES_PER_MS, SEEK_CUR);