Squash: Merge SRS 4.0

Author: winlinvip

trunk/src/app/srs_app_caster_flv.cpp

@@ -96,8 +96,9 @@ void SrsAppCasterFlv::remove(ISrsResource* c)
 srs_error_t SrsAppCasterFlv::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
 {
     SrsHttpMessage* msg = dynamic_cast<SrsHttpMessage*>(r);
-    SrsDynamicHttpConn* conn = dynamic_cast<SrsDynamicHttpConn*>(msg->connection());
-    srs_assert(conn);
+    SrsHttpConn* hconn = dynamic_cast<SrsHttpConn*>(msg->connection());
+    SrsDynamicHttpConn* dconn = dynamic_cast<SrsDynamicHttpConn*>(hconn->handler());
+    srs_assert(dconn);
 
     std::string app = srs_path_dirname(r->path());
     app = srs_string_trim_start(app, "/");
@@ -116,7 +117,7 @@ srs_error_t SrsAppCasterFlv::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessa
         o = o.substr(0, o.length() - 4);
     }
 
-    srs_error_t err = conn->proxy(w, r, o);
+    srs_error_t err = dconn->proxy(w, r, o);
     if (err != srs_success) {
         return srs_error_wrap(err, "proxy");
     }

trunk/src/app/srs_app_latest_version.cpp

@@ -51,6 +51,7 @@ void srs_build_features(stringstream& ss)
     int nn_vhosts = 0;
     bool rtsp = false, forward = false, ingest = false, edge = false, hls = false, dvr = false, flv = false;
     bool hooks = false, dash = false, hds = false, exec = false, transcode = false, security = false;
+    bool flv2 = false;
 
     SrsConfDirective* root = _srs_config->get_root();
     // Note that we limit the loop, never detect all configs.
@@ -61,6 +62,8 @@ void srs_build_features(stringstream& ss)
             string engine = _srs_config->get_stream_caster_engine(conf);
             if (engine == "rtsp") {
                 rtsp = true;
+            } else if (engine == "flv") {
+                flv2 = true;
             }
         }
 
@@ -117,6 +120,7 @@ void srs_build_features(stringstream& ss)
 
     SRS_CHECK_FEATURE2(nn_vhosts, "vhosts", ss);
     SRS_CHECK_FEATURE(rtsp, ss);
+    SRS_CHECK_FEATURE(flv2, ss);
     SRS_CHECK_FEATURE(forward, ss);
     SRS_CHECK_FEATURE(ingest, ss);
     SRS_CHECK_FEATURE(edge, ss);

trunk/src/app/srs_app_rtc_codec.cpp

@@ -10,16 +10,32 @@
 #include <srs_kernel_error.hpp>
 #include <srs_kernel_log.hpp>
 
-static const char* id2codec_name(SrsAudioCodecId id)
+static const AVCodec* srs_find_decoder_by_id(SrsAudioCodecId id)
 {
-    switch (id) {
-    case SrsAudioCodecIdAAC:
-        return "aac";
-    case SrsAudioCodecIdOpus:
-        return "libopus";
-    default:
-        return "";
+    if (id == SrsAudioCodecIdAAC) {
+        return avcodec_find_decoder_by_name("aac");
+    } else if (id == SrsAudioCodecIdOpus) {
+        const AVCodec* codec = avcodec_find_decoder_by_name("libopus");
+        if (!codec) {
+            codec = avcodec_find_decoder_by_name("opus");
+        }
+        return codec;
     }
+    return NULL;
+}
+
+static const AVCodec* srs_find_encoder_by_id(SrsAudioCodecId id)
+{
+    if (id == SrsAudioCodecIdAAC) {
+        return avcodec_find_encoder_by_name("aac");
+    } else if (id == SrsAudioCodecIdOpus) {
+        const AVCodec* codec = avcodec_find_encoder_by_name("libopus");
+        if (!codec) {
+            codec = avcodec_find_encoder_by_name("opus");
+        }
+        return codec;
+    }
+    return NULL;
 }
 
 class SrsFFmpegLogHelper {
@@ -175,10 +191,9 @@ void SrsAudioTranscoder::aac_codec_header(uint8_t **data, int *len)
 
 srs_error_t SrsAudioTranscoder::init_dec(SrsAudioCodecId src_codec)
 {
-    const char* codec_name = id2codec_name(src_codec);
-    const AVCodec *codec = avcodec_find_decoder_by_name(codec_name);
+    const AVCodec *codec = srs_find_decoder_by_id(src_codec);
     if (!codec) {
-        return srs_error_new(ERROR_RTC_RTP_MUXER, "Codec not found by name(%d,%s)", src_codec, codec_name);
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "Codec not found by %d", src_codec);
     }
 
     dec_ = avcodec_alloc_context3(codec);
@@ -208,15 +223,14 @@ srs_error_t SrsAudioTranscoder::init_dec(SrsAudioCodecId src_codec)
 
 srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_channels, int dst_samplerate, int dst_bit_rate)
 {
-    const char* codec_name = id2codec_name(dst_codec);
-    const AVCodec *codec = avcodec_find_encoder_by_name(codec_name);
+    const AVCodec *codec = srs_find_encoder_by_id(dst_codec);
     if (!codec) {
-        return srs_error_new(ERROR_RTC_RTP_MUXER, "Codec not found by name(%d,%s)", dst_codec, codec_name);
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "Codec not found by %d", dst_codec);
     }
 
     enc_ = avcodec_alloc_context3(codec);
     if (!enc_) {
-        return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio codec context(%d,%s)", dst_codec, codec_name);
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio codec context %d", dst_codec);
     }
 
     enc_->sample_rate = dst_samplerate;
@@ -228,6 +242,7 @@ srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_chan
     if (dst_codec == SrsAudioCodecIdOpus) {
         //TODO: for more level setting
         enc_->compression_level = 1;
+        enc_->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
     } else if (dst_codec == SrsAudioCodecIdAAC) {
         enc_->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
     }

trunk/src/core/srs_core_version4.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    161
+#define VERSION_REVISION    162
 
 #endif