Fix the test fail when enable ffmpeg-opus. v6.0.100 (#3868)

1. After enabling FFmpeg opus, the transcoding time for each opus packet is around 4ms. 2. To speed up case execution, our test publisher sends 400 opus packets at intervals of 1ms. 3. After the publisher starts, wait for 30ms, then the player starts. 4. Due to the lengthy processing time for each opus packet, SRS continuously receives packets from the publisher, so it doesn't switch coroutines and can't accept the player's connection. 5. Only after all opus packets are processed will it accept the player connection. Therefore, the player doesn't receive any data, leading to the failure of the case. --------- Co-authored-by: winlin <[email protected]>
ossrs · Nov 16, 2023 · 24235d8 · 24235d8
1 parent a2324a6
commit 24235d8
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Showing 8 changed files with 17 additions and 7 deletions.
diff --git a/.github/workflows/test.yml b/.github/workflows/test.yml
@@ -198,7 +198,7 @@ jobs:
       - name: Run SRS regression-test
         run: |
           docker run --rm srs:test bash -c './objs/srs -c conf/regression-test.conf && \
-              cd 3rdparty/srs-bench && (./objs/srs_test -test.v || (cat ../../objs/srs.log && exit 1)) && \
+              cd 3rdparty/srs-bench && (./objs/srs_test -test.v || (cat ../../objs/srs.log && exit 1)) && cat ../../objs/srs.log && \
               ./objs/srs_gb28181_test -test.v'
     runs-on: ubuntu-20.04
 

diff --git a/trunk/3rdparty/srs-bench/srs/rtmp_test.go b/trunk/3rdparty/srs-bench/srs/rtmp_test.go
@@ -24,7 +24,6 @@ import (
 	"bytes"
 	"context"
 	"fmt"
-	"github.com/pkg/errors"
 	"math/rand"
 	"os"
 	"sync"
@@ -36,6 +35,7 @@ import (
 	"github.com/ossrs/go-oryx-lib/logger"
 	"github.com/ossrs/go-oryx-lib/rtmp"
 	"github.com/pion/interceptor"
+	"github.com/pkg/errors"
 )
 
 func TestRtmpPublishPlay(t *testing.T) {
@@ -623,7 +623,10 @@ func TestRtmpPublish_HttpFlvPlayNoVideo(t *testing.T) {
 		go func() {
 			defer wg.Done()
 			publisher.onSendPacket = func(m *rtmp.Message) error {
-				time.Sleep(1 * time.Millisecond)
+				// Note that must greater than the cost of ffmpeg-opus, which is about 4ms, otherwise,
+				// the publisher will always get audio frames to transcode and won't accept new players
+				// connection and finally failed the case.
+				time.Sleep(5 * time.Millisecond)
 				return nil
 			}
 			if r0 = publisher.Ingest(ctx, *srsPublishAvatar); r0 != nil {

diff --git a/trunk/Dockerfile.test b/trunk/Dockerfile.test
@@ -16,7 +16,8 @@ COPY . /srs
 WORKDIR /srs/trunk
 
 # Note that we must enable the gcc7 or link failed.
-RUN ./configure --srt=on --gb28181=on --srt=on --gb28181=on --apm=on --h265=on --utest=on
+# Note that we must disable the build-cache, or it will failed, but donot know why.
+RUN ./configure --srt=on --gb28181=on --srt=on --apm=on --h265=on --utest=on --ffmpeg-opus=off --build-cache=on
 RUN make utest ${MAKEARGS}
 
 # Build benchmark tool.

diff --git a/trunk/auto/options.sh b/trunk/auto/options.sh
@@ -37,6 +37,7 @@ SRS_FFMPEG_TOOL=NO
 # FFmpeg fit is the source code for RTC, to transcode audio or video in SRS.
 SRS_FFMPEG_FIT=RESERVED
 # Whether use FFmpeg native opus codec for RTC. If not, use libopus instead.
+# Should disable it by default, because the cost of it is very high, see https://github.com/ossrs/srs/pull/3868
 SRS_FFMPEG_OPUS=NO
 # arguments
 SRS_PREFIX=/usr/local/srs

diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md
@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v6-changes"></a>
 
 ## SRS 6.0 Changelog
+* v6.0, 2023-11-16, Merge [#3868](https://github.com/ossrs/srs/pull/3868): Fix the test fail when enable ffmpeg-opus. v6.0.100 (#3868)
 * v6.0, 2023-11-15, Merge [#3879](https://github.com/ossrs/srs/pull/3879): Add --extra-ldflags. v6.0.99 (#3879)
 * v6.0, 2023-11-06, Merge [#3851](https://github.com/ossrs/srs/pull/3851): donot compile libopus when enable sys-ffmpeg. v6.0.98 (#3851)
 * v6.0, 2023-11-04, Merge [#3852](https://github.com/ossrs/srs/pull/3852): RTC: Refine FFmpeg opus audio noisy issue. v6.0.97 (#3852)

diff --git a/trunk/ide/srs_clion/CMakeLists.txt b/trunk/ide/srs_clion/CMakeLists.txt
@@ -27,11 +27,11 @@ ProcessorCount(JOBS)
 # We should always configure SRS for switching between branches.
 IF (${CMAKE_SYSTEM_NAME} MATCHES "Darwin")
     EXECUTE_PROCESS(
-        COMMAND ./configure --osx --srt=on --gb28181=on --apm=on --h265=on --utest=on --jobs=${JOBS}
+        COMMAND ./configure --osx --srt=on --gb28181=on --apm=on --h265=on --utest=on --ffmpeg-opus=off --jobs=${JOBS}
         WORKING_DIRECTORY ${SRS_DIR} RESULT_VARIABLE ret)
 ELSE ()
     EXECUTE_PROCESS(
-        COMMAND ./configure --srt=on --gb28181=on --apm=on --h265=on --utest=on --jobs=${JOBS}
+        COMMAND ./configure --srt=on --gb28181=on --apm=on --h265=on --utest=on --ffmpeg-opus=off --jobs=${JOBS}
         WORKING_DIRECTORY ${SRS_DIR} RESULT_VARIABLE ret)
 ENDIF ()
 if(NOT ret EQUAL 0)

diff --git a/trunk/src/app/srs_app_rtc_codec.cpp b/trunk/src/app/srs_app_rtc_codec.cpp
@@ -247,6 +247,10 @@ srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_chan
         //TODO: for more level setting
         enc_->compression_level = 1;
         enc_->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+
+#ifdef SRS_FFMPEG_OPUS
+        av_opt_set(enc_->priv_data, "opus_delay", "2.5", 0);
+#endif
     } else if (dst_codec == SrsAudioCodecIdAAC) {
         enc_->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
     }

diff --git a/trunk/src/core/srs_core_version6.hpp b/trunk/src/core/srs_core_version6.hpp
@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       6
 #define VERSION_MINOR       0
-#define VERSION_REVISION    99
+#define VERSION_REVISION    100
 
 #endif