refine hls codec sample info. 0.9.161

@@ -30,26 +30,14 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <srs_kernel_codec.hpp>
 #include <srs_kernel_stream.hpp>
 
-SrsCodecBuffer::SrsCodecBuffer()
+SrsCodecSampleUnit::SrsCodecSampleUnit()
 {
     size = 0;
     bytes = NULL;
 }
 
-void SrsCodecBuffer::append(void* data, int len)
+SrsCodecSampleUnit::~SrsCodecSampleUnit()
 {
-    srs_assert(data);
-    srs_assert(len > 0);
-
-    bytes = (char*)realloc(bytes, size + len);
-    memcpy(bytes + size, data, len);
-    size += len;
-}
-
-void SrsCodecBuffer::free()
-{
-    size = 0;
-    srs_freep(bytes);
 }
 
 SrsCodecSample::SrsCodecSample()
@@ -64,7 +52,7 @@ SrsCodecSample::~SrsCodecSample()
 void SrsCodecSample::clear()
 {
     is_video = false;
-    nb_buffers = 0;
+    nb_sample_units = 0;
 
     cts = 0;
     frame_type = SrsCodecVideoAVCFrameReserved;
@@ -76,20 +64,20 @@ void SrsCodecSample::clear()
     aac_packet_type = SrsCodecAudioTypeReserved;
 }
 
-int SrsCodecSample::add_sample(char* bytes, int size)
+int SrsCodecSample::add_sample_unit(char* bytes, int size)
 {
     int ret = ERROR_SUCCESS;
     
-    if (nb_buffers >= SRS_MAX_CODEC_SAMPLE) {
+    if (nb_sample_units >= SRS_MAX_CODEC_SAMPLE) {
         ret = ERROR_HLS_DECODE_ERROR;
         srs_error("hls decode samples error, "
             "exceed the max count: %d, ret=%d", SRS_MAX_CODEC_SAMPLE, ret);
         return ret;
     }
     
-    SrsCodecBuffer* buf = &buffers[nb_buffers++];
-    buf->bytes = bytes;
-    buf->size = size;
+    SrsCodecSampleUnit* sample_unit = &sample_units[nb_sample_units++];
+    sample_unit->bytes = bytes;
+    sample_unit->size = size;
     
     return ret;
 }
@@ -239,7 +227,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
         
         // Raw AAC frame data in UI8 []
         // 6.3 Raw Data, aac-iso-13818-7.pdf, page 28
-        if ((ret = sample->add_sample(stream->data() + stream->pos(), stream->size() - stream->pos())) != ERROR_SUCCESS) {
+        if ((ret = sample->add_sample_unit(stream->data() + stream->pos(), stream->size() - stream->pos())) != ERROR_SUCCESS) {
             srs_error("hls add audio sample failed. ret=%d", ret);
             return ret;
         }
@@ -445,7 +433,7 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
                 return ret;
             }
             // 7.3.1 NAL unit syntax, H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
-            if ((ret = sample->add_sample(stream->data() + stream->pos(), NALUnitLength)) != ERROR_SUCCESS) {
+            if ((ret = sample->add_sample_unit(stream->data() + stream->pos(), NALUnitLength)) != ERROR_SUCCESS) {
                 srs_error("hls add video sample failed. ret=%d", ret);
                 return ret;
             }

@@ -89,30 +89,38 @@ enum SrsCodecAudioSoundType
 };
 
 /**
-* buffer indicates the position and size.
+* the codec sample unit.
+* for h.264 video packet, a NALU is a sample unit.
+* for aac raw audio packet, a NALU is the entire aac raw data.
+* for sequence header, it's not a sample unit.
 */
-class SrsCodecBuffer
+class SrsCodecSampleUnit
 {
 public:
     /**
-    * @remark user must manage the bytes.
+    * the sample bytes is directly ptr to packet bytes,
+    * user should never use it when packet destroyed.
     */
     int size;
     char* bytes;
 public:
-    SrsCodecBuffer();
-    void append(void* data, int len);
-public:
-    /**
-    * free the bytes, 
-    * user can invoke it to free the bytes,
-    * the SrsCodecBuffer never free automatically.
-    */
-    void free();
+    SrsCodecSampleUnit();
+    virtual ~SrsCodecSampleUnit();
 };
 
 /**
 * the samples in the flv audio/video packet.
+* the sample used to analysis a video/audio packet,
+* split the h.264 NALUs to buffers, or aac raw data to a buffer,
+* and decode the video/audio specified infos.
+* 
+* the sample unit:
+*       a video packet codec in h.264 contains many NALUs, each is a sample unit.
+*       a audio packet codec in aac is a sample unit.
+* @remark, the video/audio sequence header is not sample unit,
+*       all sequence header stores as extra data, 
+*       @see SrsAvcAacCodec.avc_extra_data and SrsAvcAacCodec.aac_extra_data
+* @remark, user must clear all samples before decode a new video/audio packet.
 */
 class SrsCodecSample
 {
@@ -123,12 +131,14 @@ public:
     * generally, aac audio packet corresponding to one buffer,
     * where avc/h264 video packet may contains multiple buffer.
     */
-    int nb_buffers;
-    SrsCodecBuffer buffers[SRS_MAX_CODEC_SAMPLE];
+    int nb_sample_units;
+    SrsCodecSampleUnit sample_units[SRS_MAX_CODEC_SAMPLE];
 public:
     bool is_video;
-    // CompositionTime, video_file_format_spec_v10_1.pdf, page 78.
-    // cts = pts - dts, where dts = flvheader->timestamp.
+    /**
+    * CompositionTime, video_file_format_spec_v10_1.pdf, page 78.
+    * cts = pts - dts, where dts = flvheader->timestamp.
+    */
     int32_t cts;
 public:
     // video specified
@@ -143,13 +153,37 @@ public:
 public:
     SrsCodecSample();
     virtual ~SrsCodecSample();
+public:
+    /**
+    * clear all samples.
+    * the sample units never copy the bytes, it directly use the ptr,
+    * so when video/audio packet is destroyed, the sample must be clear.
+    * in a word, user must clear sample before demux it.
+    * @remark demux sample use SrsAvcAacCodec.audio_aac_demux or video_avc_demux.
+    */
     void clear();
-    int add_sample(char* bytes, int size);
+    /**
+    * add the a sample unit, it's a h.264 NALU or aac raw data.
+    * the sample unit directly use the ptr of packet bytes,
+    * so user must never use sample unit when packet is destroyed.
+    * in a word, user must clear sample before demux it.
+    */
+    int add_sample_unit(char* bytes, int size);
 };
 
 /**
 * the h264/avc and aac codec, for media stream.
-* to decode the stream of avc/aac for hls.
+*
+* to demux the FLV/RTMP video/audio packet to sample,
+* add each NALUs of h.264 as a sample unit to sample,
+* while the entire aac raw data as a sample unit.
+*
+* for sequence header,
+* demux it and save it in the avc_extra_data and aac_extra_data,
+* 
+* for the codec info, such as audio sample rate,
+* decode from FLV/RTMP header, then use codec info in sequence 
+* header to override it.
 */
 class SrsAvcAacCodec
 {

@@ -68,6 +68,7 @@ using namespace std;
 #include <srs_kernel_utility.hpp>
 #include <srs_app_avc_aac.hpp>
 #include <srs_kernel_file.hpp>
+#include <srs_kernel_buffer.hpp>
 
 // max PES packets size to flush the video.
 #define SRS_AUTO_HLS_AUDIO_CACHE_SIZE 1024 * 1024
@@ -195,16 +196,16 @@ public:
 
         return ret;
     }
-    static int write_frame(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsCodecBuffer* buffer)
+    static int write_frame(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsBuffer* buffer)
     {
         int ret = ERROR_SUCCESS;
         
-        if (!buffer->bytes || buffer->size <= 0) {
+        if (!buffer->bytes() || buffer->length() <= 0) {
             return ret;
         }
         
-        char* last = buffer->bytes + buffer->size;
-        char* pos = buffer->bytes;
+        char* last = buffer->bytes() + buffer->length();
+        char* pos = buffer->bytes();
         
         bool first = true;
         while (pos < last) {
@@ -478,7 +479,7 @@ int SrsTSMuxer::open(string _path)
     return ret;
 }
 
-int SrsTSMuxer::write_audio(SrsMpegtsFrame* af, SrsCodecBuffer* ab)
+int SrsTSMuxer::write_audio(SrsMpegtsFrame* af, SrsBuffer* ab)
 {
     int ret = ERROR_SUCCESS;
     
@@ -489,7 +490,7 @@ int SrsTSMuxer::write_audio(SrsMpegtsFrame* af, SrsCodecBuffer* ab)
     return ret;
 }
 
-int SrsTSMuxer::write_video(SrsMpegtsFrame* vf, SrsCodecBuffer* vb)
+int SrsTSMuxer::write_video(SrsMpegtsFrame* vf, SrsBuffer* vb)
 {
     int ret = ERROR_SUCCESS;
     
@@ -651,7 +652,7 @@ bool SrsHlsMuxer::is_segment_overflow()
     return current->duration >= hls_fragment;
 }
 
-int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsCodecBuffer* ab)
+int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsBuffer* ab)
 {
     int ret = ERROR_SUCCESS;
 
@@ -661,7 +662,7 @@ int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsCodecBuffer* ab)
         return ret;
     }
     
-    if (ab->size <= 0) {
+    if (ab->length() <= 0) {
         return ret;
     }
     
@@ -673,13 +674,12 @@ int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsCodecBuffer* ab)
     }
     
     // write success, clear and free the buffer
-    ab->free();
+    ab->erase(ab->length());
 
     return ret;
 }
 
-int SrsHlsMuxer::flush_video(
-    SrsMpegtsFrame* af, SrsCodecBuffer* ab, SrsMpegtsFrame* vf, SrsCodecBuffer* vb)
+int SrsHlsMuxer::flush_video(SrsMpegtsFrame* af, SrsBuffer* ab, SrsMpegtsFrame* vf, SrsBuffer* vb)
 {
     int ret = ERROR_SUCCESS;
 
@@ -699,7 +699,7 @@ int SrsHlsMuxer::flush_video(
     }
     
     // write success, clear and free the buffer
-    vb->free();
+    vb->erase(vb->length());
     
     return ret;
 }
@@ -962,8 +962,8 @@ SrsHlsCache::SrsHlsCache()
 {
     aac_jitter = new SrsHlsAacJitter();
     
-    ab = new SrsCodecBuffer();
-    vb = new SrsCodecBuffer();
+    ab = new SrsBuffer();
+    vb = new SrsBuffer();
     
     af = new SrsMpegtsFrame();
     vf = new SrsMpegtsFrame();
@@ -975,8 +975,8 @@ SrsHlsCache::~SrsHlsCache()
 {
     srs_freep(aac_jitter);
     
-    ab->free();
-    vb->free();
+    ab->erase(ab->length());
+    vb->erase(vb->length());
     
     srs_freep(ab);
     srs_freep(vb);
@@ -1051,7 +1051,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     int ret = ERROR_SUCCESS;
     
     // start buffer, set the af
-    if (ab->size == 0) {
+    if (ab->length() == 0) {
         pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
         
         af->dts = af->pts = audio_buffer_start_pts = pts;
@@ -1067,7 +1067,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     }
     
     // flush if buffer exceed max size.
-    if (ab->size > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
+    if (ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
         if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
             return ret;
         }
@@ -1159,11 +1159,11 @@ int SrsHlsCache::cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample)
 {
     int ret = ERROR_SUCCESS;
     
-    for (int i = 0; i < sample->nb_buffers; i++) {
-        SrsCodecBuffer* buf = &sample->buffers[i];
-        int32_t size = buf->size;
+    for (int i = 0; i < sample->nb_sample_units; i++) {
+        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
+        int32_t size = sample_unit->size;
         
-        if (!buf->bytes || size <= 0 || size > 0x1fff) {
+        if (!sample_unit->bytes || size <= 0 || size > 0x1fff) {
             ret = ERROR_HLS_AAC_FRAME_LENGTH;
             srs_error("invalid aac frame length=%d, ret=%d", size, ret);
             return ret;
@@ -1215,8 +1215,8 @@ int SrsHlsCache::cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample)
         adts_header[5] |= 0x1f;
 
         // copy to audio buffer
-        ab->append(adts_header, sizeof(adts_header));
-        ab->append(buf->bytes, buf->size);
+        ab->append((const char*)adts_header, sizeof(adts_header));
+        ab->append(sample_unit->bytes, sample_unit->size);
     }
     
     return ret;
@@ -1239,11 +1239,11 @@ int SrsHlsCache::cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample)
     *       xxxxxxx // data bytes.
     * so, for each sample, we append header in aud_nal, then appends the bytes in sample.
     */
-    for (int i = 0; i < sample->nb_buffers; i++) {
-        SrsCodecBuffer* buf = &sample->buffers[i];
-        int32_t size = buf->size;
+    for (int i = 0; i < sample->nb_sample_units; i++) {
+        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
+        int32_t size = sample_unit->size;
         
-        if (!buf->bytes || size <= 0) {
+        if (!sample_unit->bytes || size <= 0) {
             ret = ERROR_HLS_AVC_SAMPLE_SIZE;
             srs_error("invalid avc sample length=%d, ret=%d", size, ret);
             return ret;
@@ -1259,7 +1259,7 @@ int SrsHlsCache::cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample)
         // 5bits, 7.3.1 NAL unit syntax, 
         // H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
         u_int8_t nal_unit_type;
-        nal_unit_type = *buf->bytes;
+        nal_unit_type = *sample_unit->bytes;
         nal_unit_type &= 0x1f;
         
         // @see: ngx_rtmp_hls_video
@@ -1277,7 +1277,7 @@ int SrsHlsCache::cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample)
             }
             if (nal_unit_type == 1 || nal_unit_type == 5 || nal_unit_type == 6) {
                 // for type 6, append a aud with type 9.
-                vb->append(aud_nal, sizeof(aud_nal));
+                vb->append((const char*)aud_nal, sizeof(aud_nal));
                 aud_sent = true;
             }
         }
@@ -1290,13 +1290,13 @@ int SrsHlsCache::cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample)
             // @see: ngx_rtmp_hls_append_sps_pps
             if (codec->sequenceParameterSetLength > 0) {
                 // AnnexB prefix, for sps always 4 bytes header
-                vb->append(aud_nal, 4);
+                vb->append((const char*)aud_nal, 4);
                 // sps
                 vb->append(codec->sequenceParameterSetNALUnit, codec->sequenceParameterSetLength);
             }
             if (codec->pictureParameterSetLength > 0) {
                 // AnnexB prefix, for pps always 4 bytes header
-                vb->append(aud_nal, 4);
+                vb->append((const char*)aud_nal, 4);
                 // pps
                 vb->append(codec->pictureParameterSetNALUnit, codec->pictureParameterSetLength);
             }
@@ -1318,13 +1318,13 @@ int SrsHlsCache::cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample)
         u_int8_t* end = p + 3;
         
         // first AnnexB prefix is long (4 bytes)
-        if (vb->size == 0) {
+        if (vb->length() == 0) {
             p = aud_nal;
         }
-        vb->append(p, end - p);
+        vb->append((const char*)p, end - p);
         
         // sample data
-        vb->append(buf->bytes, buf->size);
+        vb->append(sample_unit->bytes, sample_unit->size);
     }
     
     return ret;

@@ -50,9 +50,9 @@ extern int aac_sample_rates[];
 #include <string>
 #include <vector>
 
+class SrsBuffer;
 class SrsSharedPtrMessage;
 class SrsCodecSample;
-class SrsCodecBuffer;
 class SrsMpegtsFrame;
 class SrsAmf0Object;
 class SrsRtmpJitter;
@@ -109,8 +109,8 @@ public:
     virtual ~SrsTSMuxer();
 public:
     virtual int open(std::string _path);
-    virtual int write_audio(SrsMpegtsFrame* af, SrsCodecBuffer* ab);
-    virtual int write_video(SrsMpegtsFrame* vf, SrsCodecBuffer* vb);
+    virtual int write_audio(SrsMpegtsFrame* af, SrsBuffer* ab);
+    virtual int write_video(SrsMpegtsFrame* vf, SrsBuffer* vb);
     virtual void close();
 };
 
@@ -196,8 +196,8 @@ public:
     * that is whether the current segment duration >= the segment in config
     */
     virtual bool is_segment_overflow();
-    virtual int flush_audio(SrsMpegtsFrame* af, SrsCodecBuffer* ab);
-    virtual int flush_video(SrsMpegtsFrame* af, SrsCodecBuffer* ab, SrsMpegtsFrame* vf, SrsCodecBuffer* vb);
+    virtual int flush_audio(SrsMpegtsFrame* af, SrsBuffer* ab);
+    virtual int flush_video(SrsMpegtsFrame* af, SrsBuffer* ab, SrsMpegtsFrame* vf, SrsBuffer* vb);
     /**
     * close segment(ts).
     * @param log_desc the description for log.
@@ -231,9 +231,9 @@ class SrsHlsCache
 private:
     // current frame and buffer
     SrsMpegtsFrame* af;
-    SrsCodecBuffer* ab;
+    SrsBuffer* ab;
     SrsMpegtsFrame* vf;
-    SrsCodecBuffer* vb;
+    SrsBuffer* vb;
 private:
     // the audio cache buffer start pts, to flush audio if full.
     int64_t audio_buffer_start_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 "160"
+#define VERSION_REVISION "161"
 #define RTMP_SIG_SRS_VERSION VERSION_MAJOR"."VERSION_MINOR"."VERSION_REVISION
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"