fix #81: support all aac sample rate, for instance, 48000. to 0.9.150

@@ -108,7 +108,7 @@ SrsAvcAacCodec::SrsAvcAacCodec()
     avc_profile             = 0;
     avc_level                 = 0;
     aac_profile                = 0;
-    aac_sample_rate            = 0;
+    aac_sample_rate            = _SRS_AAC_SAMPLE_RATE_UNSET; // sample rate ignored
     aac_channels            = 0;
     avc_extra_size             = 0;
     avc_extra_data             = NULL;
@@ -166,25 +166,6 @@ int SrsAvcAacCodec::audio_aac_demux(int8_t* data, int size, SrsCodecSample* samp
     sample->sound_rate = (SrsCodecAudioSampleRate)sound_rate;
     sample->sound_size = (SrsCodecAudioSampleSize)sound_size;
     
-    // reset the sample rate by sequence header
-    static int aac_sample_rates[] = {
-        96000, 88200, 64000, 48000,
-        44100, 32000, 24000, 22050,
-        16000, 12000, 11025,  8000,
-        7350,     0,     0,    0
-    };
-    switch (aac_sample_rates[aac_sample_rate]) {
-        case 11025:
-            sample->sound_rate = SrsCodecAudioSampleRate11025;
-            break;
-        case 22050:
-            sample->sound_rate = SrsCodecAudioSampleRate22050;
-            break;
-        case 44100:
-            sample->sound_rate = SrsCodecAudioSampleRate44100;
-            break;
-    };
-    
     // only support aac
     if (audio_codec_id != SrsCodecAudioAAC) {
         ret = ERROR_HLS_DECODE_ERROR;
@@ -261,6 +242,27 @@ int SrsAvcAacCodec::audio_aac_demux(int8_t* data, int size, SrsCodecSample* samp
         // ignored.
     }
     
+    // reset the sample rate by sequence header
+    if (aac_sample_rate != _SRS_AAC_SAMPLE_RATE_UNSET) {
+        static int aac_sample_rates[] = {
+            96000, 88200, 64000, 48000,
+            44100, 32000, 24000, 22050,
+            16000, 12000, 11025,  8000,
+            7350,     0,     0,    0
+        };
+        switch (aac_sample_rates[aac_sample_rate]) {
+            case 11025:
+                sample->sound_rate = SrsCodecAudioSampleRate11025;
+                break;
+            case 22050:
+                sample->sound_rate = SrsCodecAudioSampleRate22050;
+                break;
+            case 44100:
+                sample->sound_rate = SrsCodecAudioSampleRate44100;
+                break;
+        };
+    }
+    
     srs_info("audio decoded, type=%d, codec=%d, asize=%d, rate=%d, format=%d, size=%d", 
         sound_type, audio_codec_id, sound_size, sound_rate, sound_format, size);
     

@@ -35,6 +35,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 class SrsStream;
 
 #define SRS_MAX_CODEC_SAMPLE 128
+#define _SRS_AAC_SAMPLE_RATE_UNSET 15
 
 // Sampling rate. The following values are defined:
 // 0 = 5.5 kHz = 5512 Hz

@@ -365,7 +365,7 @@ SrsHlsAacJitter::~SrsHlsAacJitter()
 {
 }
 
-int64_t SrsHlsAacJitter::on_buffer_start(int64_t flv_pts, int sample_rate)
+int64_t SrsHlsAacJitter::on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate)
 {
     // 0 = 5.5 kHz = 5512 Hz
     // 1 = 11 kHz = 11025 Hz
@@ -374,6 +374,17 @@ int64_t SrsHlsAacJitter::on_buffer_start(int64_t flv_pts, int sample_rate)
     static int flv_sample_rates[] = {5512, 11025, 22050, 44100};
     int flv_sample_rate = flv_sample_rates[sample_rate & 0x03];
 
+    // reset the sample rate by sequence header
+    if (aac_sample_rate != _SRS_AAC_SAMPLE_RATE_UNSET) {
+        static int aac_sample_rates[] = {
+            96000, 88200, 64000, 48000,
+            44100, 32000, 24000, 22050,
+            16000, 12000, 11025,  8000,
+            7350,     0,     0,    0
+        };
+        flv_sample_rate = aac_sample_rates[aac_sample_rate];
+    }
+
     // sync time set to 0, donot adjust the aac timestamp.
     if (!sync_ms) {
         return flv_pts;
@@ -1016,7 +1027,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     
     // start buffer, set the af
     if (ab->size == 0) {
-        pts = aac_jitter->on_buffer_start(pts, sample->sound_rate);
+        pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
         
         af->dts = af->pts = audio_buffer_start_pts = pts;
         af->pid = TS_AUDIO_PID;

@@ -66,9 +66,11 @@ public:
     /**
     * when buffer start, calc the "correct" pts for ts,
     * @param flv_pts, the flv pts calc from flv header timestamp,
+    * @param sample_rate, the sample rate in format(flv/RTMP packet header).
+    * @param aac_sample_rate, the sample rate in codec(sequence header).
     * @return the calc correct pts.
     */
-    virtual int64_t on_buffer_start(int64_t flv_pts, int sample_rate);
+    virtual int64_t on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate);
     /**
     * when buffer continue, muxer donot write to file,
     * the audio buffer continue grow and donot need a pts,

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 #define VERSION_MAJOR "0"
 #define VERSION_MINOR "9"
-#define VERSION_REVISION "149"
+#define VERSION_REVISION "150"
 #define RTMP_SIG_SRS_VERSION VERSION_MAJOR"."VERSION_MINOR"."VERSION_REVISION
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"

@@ -33,6 +33,7 @@ using namespace std;
 
 SrsFileWriter::SrsFileWriter()
 {
+    fd = -1;
 }
 
 SrsFileWriter::~SrsFileWriter()