fix RTMP publish single AAC from ffmpeg client.

harlanc · Mar 28, 2024 · 437d489 · 437d489
1 parent d3a9bbf
commit 437d489
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Showing 4 changed files with 28 additions and 6 deletions.
diff --git a/library/container/flv/src/mpeg4_aac.rs b/library/container/flv/src/mpeg4_aac.rs
@@ -436,7 +436,17 @@ impl Mpeg4AacProcessor {
             0xF0 /* 12-syncword */ | (id << 3)/*1-ID*/| 0x01, /*1-protection_absent*/
         )?; //1
 
-        let profile = self.mpeg4_aac.object_type;
+        // workaround: profile - 1 overflow error. If profile == 0:reserved, then force
+        // profile to 2:AAC_LC.
+        // If publish pure AAC file to RTMP by ffmpeg, the AAC object type is 0:reserved,
+        // If publish a opus file to RTMP, ffmpeg will convert opus to aac, then use FLV as container,
+        // the object type is normal 2:AAC_LC.
+        let profile = if self.mpeg4_aac.object_type > 0 {
+            self.mpeg4_aac.object_type
+        } else {
+            2
+        };
+
         let sampling_frequency_index = self.mpeg4_aac.sampling_frequency_index;
         let channel_configuration = self.mpeg4_aac.channel_configuration;
         self.bytes_writer.write_u8(

diff --git a/library/streamhub/src/lib.rs b/library/streamhub/src/lib.rs
@@ -84,6 +84,11 @@ impl StreamDataTransceiver {
                     data: _,
                 } => {}
                 FrameData::Audio { timestamp, data } => {
+                    if data.len() <= 4 {
+                        log::info!("ignore received Audio Frame with len {}", data.len());
+                        return;
+                    }
+
                     let data = FrameData::Audio {
                         timestamp,
                         data: data.clone(),

diff --git a/protocol/hls/src/flv2hls.rs b/protocol/hls/src/flv2hls.rs
@@ -67,8 +67,13 @@ impl Flv2HlsRemuxer {
     pub fn process_flv_data(&mut self, data: FlvData) -> Result<(), MediaError> {
         let flv_demux_data: FlvDemuxerData = match data {
             FlvData::Audio { timestamp, data } => {
-                let audio_data = self.audio_demuxer.demux(timestamp, data)?;
-                FlvDemuxerData::Audio { data: audio_data }
+                if data.len() > 4 {
+                    // log::error!("process_flv_data len={}", data.len());
+                    let audio_data = self.audio_demuxer.demux(timestamp, data)?;
+                    FlvDemuxerData::Audio { data: audio_data }
+                } else {
+                    return Ok(());
+                }
             }
             FlvData::Video { timestamp, data } => {
                 if let Some(video_data) = self.video_demuxer.demux(timestamp, data)? {

diff --git a/protocol/rtmp/src/cache/mod.rs b/protocol/rtmp/src/cache/mod.rs
@@ -33,7 +33,6 @@ pub struct Cache {
 
 impl Cache {
     pub fn new(gop_num: usize, statistic_data_sender: Option<StatisticDataSender>) -> Self {
-
         Cache {
             metadata: metadata::MetaData::new(),
             metadata_timestamp: 0,
@@ -78,7 +77,10 @@ impl Cache {
         let mut reader = BytesReader::new(chunk_body.clone());
         let tag_header = AudioTagHeader::unmarshal(&mut reader)?;
 
-        if tag_header.sound_format == define::SoundFormat::AAC as u8
+        let remain_bytes = reader.extract_remaining_bytes();
+
+        if remain_bytes.len() >= 2
+            && tag_header.sound_format == define::SoundFormat::AAC as u8
             && tag_header.aac_packet_type == define::aac_packet_type::AAC_SEQHDR
         {
             self.audio_seq = chunk_body.clone();
@@ -88,7 +90,7 @@ impl Cache {
                 let mut aac_processor = Mpeg4AacProcessor::default();
 
                 let aac = aac_processor
-                    .extend_data(reader.extract_remaining_bytes())
+                    .extend_data(remain_bytes)
                     .audio_specific_config_load()?;
 
                 let statistic_audio_codec = StatisticData::AudioCodec {