Merge pull request #1636 from runner365/3.0release.srt.dev

support all aac sample rate in srt2rtmp

trunk/src/protocol/srs_raw_avc.cpp

@@ -438,7 +438,8 @@ srs_error_t SrsRawAacStream::adts_demux(SrsBuffer* stream, char** pframe, int* p
 srs_error_t SrsRawAacStream::mux_sequence_header(SrsRawAacStreamCodec* codec, string& sh)
 {
     srs_error_t err = srs_success;
-    
+    char samplingFrequencyIndex = codec->sampling_frequency_index;
+
     // only support aac profile 1-4.
     if (codec->aac_object == SrsAacObjectTypeReserved) {
         return srs_error_new(ERROR_AAC_DATA_INVALID, "invalid aac object");
@@ -446,21 +447,11 @@ srs_error_t SrsRawAacStream::mux_sequence_header(SrsRawAacStreamCodec* codec, st
     
     SrsAacObjectType audioObjectType = codec->aac_object;
     char channelConfiguration = codec->channel_configuration;
-    char samplingFrequencyIndex = codec->sampling_frequency_index;
     
-    // override the aac samplerate by user specified.
-    // @see https://github.com/ossrs/srs/issues/212#issuecomment-64146899
-    switch (codec->sound_rate) {
-        case SrsAudioSampleRate11025:
-            samplingFrequencyIndex = 0x0a; break;
-        case SrsAudioSampleRate22050:
-            samplingFrequencyIndex = 0x07; break;
-        case SrsAudioSampleRate44100:
-            samplingFrequencyIndex = 0x04; break;
-        default:
-            break;
+    if (samplingFrequencyIndex >= 16) {
+        samplingFrequencyIndex = 4;//default 44100
     }
-    
+
     char chs[2];
     // @see ISO_IEC_14496-3-AAC-2001.pdf
     // AudioSpecificConfig (), page 33

trunk/src/protocol/srs_raw_avc.hpp

@@ -30,6 +30,11 @@
 
 #include <srs_kernel_codec.hpp>
 
+static const int mpeg4audio_sample_rates[16] = {
+    96000, 88200, 64000, 48000, 44100, 32000,
+    24000, 22050, 16000, 12000, 11025, 8000, 7350
+};
+
 class SrsBuffer;
 
 // The raw h.264 stream, in annexb.

trunk/src/srt/srt_to_rtmp.cpp

@@ -517,25 +517,11 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
     return err;
 }
 
-int rtmp_client::get_sample_rate(char sound_rate) {
+int rtmp_client::get_sample_rate(char sample_index) {
     int sample_rate = 44100;
 
-    switch (sound_rate)
-    {
-    case SrsAudioSampleRate44100:
-        sample_rate = 44100;
-        break;
-    case SrsAudioSampleRate22050:
-        sample_rate = 22050;
-        break;
-    case SrsAudioSampleRate11025:
-        sample_rate = 11025;
-        break;
-    case SrsAudioSampleRate5512:
-        sample_rate = 5512;
-        break;
-    default:
-        break;
+    if ((sample_index >= 0) && (sample_index < 16)) {
+        sample_rate = mpeg4audio_sample_rates[sample_index];
     }
     return sample_rate;
 }