From f1db76011bb1a4075de835a66c99555ee5887598 Mon Sep 17 00:00:00 2001 From: chundonglinlin Date: Sat, 4 Nov 2023 16:21:44 +0800 Subject: [PATCH] RTC: Refine FFmpeg opus audio noisy issue. v5.0.197 (#3852) When converting between AAC and Opus formats (aac2opus or opus2aac), the `av_frame_get_buffer` API is frequently called. The goal is to optimize the code logic and reduce the frequent allocation and deallocation of memory. In the case of aac2opus, av_frame_get_buffer is still frequently called. In the case of opus2aac, the goal is to avoid calling av_frame_get_buffer and reduce memory allocations. Before calling the `av_audio_fifo_read` API, use `av_frame_make_writable` to check if the frame is writable. If it is not writable, create a new frame. --------- Co-authored-by: john --- .github/workflows/test.yml | 3 ++- trunk/auto/options.sh | 3 +-- trunk/doc/CHANGELOG.md | 1 + trunk/research/players/js/srs.page.js | 1 - trunk/research/players/srs_player.html | 6 +++--- trunk/research/players/whep.html | 6 +++--- trunk/research/players/whip.html | 6 +++--- trunk/src/app/srs_app_rtc_codec.cpp | 20 +++++++++++--------- trunk/src/core/srs_core_version5.hpp | 2 +- 9 files changed, 25 insertions(+), 23 deletions(-) diff --git a/.github/workflows/test.yml b/.github/workflows/test.yml index 621ba7569d..65875cb5a8 100644 --- a/.github/workflows/test.yml +++ b/.github/workflows/test.yml @@ -195,7 +195,8 @@ 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 && ./objs/srs_gb28181_test -test.v' + cd 3rdparty/srs-bench && (./objs/srs_test -test.v || (cat ../../objs/srs.log && exit 1)) && \ + ./objs/srs_gb28181_test -test.v' runs-on: ubuntu-20.04 coverage: diff --git a/trunk/auto/options.sh b/trunk/auto/options.sh index 2f8c7dad37..2f97baee94 100755 --- a/trunk/auto/options.sh +++ b/trunk/auto/options.sh @@ -448,8 +448,7 @@ function parse_user_option() { --ffmpeg-tool) SRS_FFMPEG_TOOL=$(switch2value $value) ;; # use cache for build. - --build-cache) SRS_BUILD_CACHE=YES ;; - --without-build-cache) SRS_BUILD_CACHE=NO ;; + --build-cache) SRS_BUILD_CACHE=$(switch2value $value) ;; *) echo "$0: error: invalid option \"$option\"" diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md index b9cf22414c..99e21db91c 100644 --- a/trunk/doc/CHANGELOG.md +++ b/trunk/doc/CHANGELOG.md @@ -7,6 +7,7 @@ The changelog for SRS. ## SRS 5.0 Changelog +* v5.0, 2023-11-04, Merge [#3852](https://github.com/ossrs/srs/pull/3852): RTC: Refine FFmpeg opus audio noisy issue. v5.0.197 (#3852) * v5.0, 2023-11-01, Merge [#3858](https://github.com/ossrs/srs/pull/3858): Support build without cache to test if actions fail. v5.0.196 (#3858) * v5.0, 2023-10-25, Merge [#3845](https://github.com/ossrs/srs/pull/3845): RTC: Fix FFmpeg opus audio noisy issue. v5.0.195 (#3845) * v5.0, 2023-10-21, Merge [#3847](https://github.com/ossrs/srs/pull/3847): WebRTC: TCP transport should use read_fully instead of read. v5.0.194 (#3847) diff --git a/trunk/research/players/js/srs.page.js b/trunk/research/players/js/srs.page.js index ff4415445b..520491e48d 100755 --- a/trunk/research/players/js/srs.page.js +++ b/trunk/research/players/js/srs.page.js @@ -13,7 +13,6 @@ function srs_get_player_height() { return srs_get_player_width() * 9 / 19; } * update the navigator, add same query string. */ function update_nav() { - $("#srs_index").attr("href", "index.html" + window.location.search); $("#nav_srs_player").attr("href", "srs_player.html" + window.location.search); $("#nav_rtc_player").attr("href", "rtc_player.html" + window.location.search); $("#nav_rtc_publisher").attr("href", "rtc_publisher.html" + window.location.search); diff --git a/trunk/research/players/srs_player.html b/trunk/research/players/srs_player.html index 8ce4b616e0..40bc042404 100755 --- a/trunk/research/players/srs_player.html +++ b/trunk/research/players/srs_player.html @@ -15,7 +15,7 @@
- SRS + SRS
diff --git a/trunk/research/players/whep.html b/trunk/research/players/whep.html index ead5018fdf..af896da774 100644 --- a/trunk/research/players/whep.html +++ b/trunk/research/players/whep.html @@ -20,7 +20,7 @@
- SRS + SRS
diff --git a/trunk/research/players/whip.html b/trunk/research/players/whip.html index cff306656c..4f804621f1 100644 --- a/trunk/research/players/whip.html +++ b/trunk/research/players/whip.html @@ -20,7 +20,7 @@
- SRS + SRS
diff --git a/trunk/src/app/srs_app_rtc_codec.cpp b/trunk/src/app/srs_app_rtc_codec.cpp index cbc46d1388..c3db8e7b70 100644 --- a/trunk/src/app/srs_app_rtc_codec.cpp +++ b/trunk/src/app/srs_app_rtc_codec.cpp @@ -260,6 +260,13 @@ srs_error_t SrsAudioTranscoder::init_enc(SrsAudioCodecId dst_codec, int dst_chan if (!enc_frame_) { return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not allocate audio encode in frame"); } + enc_frame_->format = enc_->sample_fmt; + enc_frame_->nb_samples = enc_->frame_size; + enc_frame_->channel_layout = enc_->channel_layout; + + if (av_frame_get_buffer(enc_frame_, 0) < 0) { + return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not get audio frame buffer"); + } enc_packet_ = av_packet_alloc(); if (!enc_packet_) { @@ -381,26 +388,21 @@ srs_error_t SrsAudioTranscoder::encode(std::vector &pkts) } while (av_audio_fifo_size(fifo_) >= enc_->frame_size) { - enc_frame_->format = enc_->sample_fmt; - enc_frame_->nb_samples = enc_->frame_size; - enc_frame_->channel_layout = enc_->channel_layout; - - if (av_frame_get_buffer(enc_frame_, 0) < 0) { - av_frame_free(&enc_frame_); - return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not get audio frame buffer"); + // make sure the frame is writable + if (av_frame_make_writable(enc_frame_) < 0) { + return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not make writable frame"); } /* Read as many samples from the FIFO buffer as required to fill the frame. * The samples are stored in the frame temporarily. */ if (av_audio_fifo_read(fifo_, (void **)enc_frame_->data, enc_->frame_size) < enc_->frame_size) { - av_frame_free(&enc_frame_); return srs_error_new(ERROR_RTC_RTP_MUXER, "Could not read data from FIFO"); } + /* send the frame for encoding */ enc_frame_->pts = next_out_pts_; next_out_pts_ += enc_->frame_size; int error = avcodec_send_frame(enc_, enc_frame_); - av_frame_unref(enc_frame_); if (error < 0) { return srs_error_new(ERROR_RTC_RTP_MUXER, "Error sending the frame to the encoder(%d,%s)", error, av_make_error_string(err_buf, AV_ERROR_MAX_STRING_SIZE, error)); diff --git a/trunk/src/core/srs_core_version5.hpp b/trunk/src/core/srs_core_version5.hpp index 554b2f70b0..014b230324 100644 --- a/trunk/src/core/srs_core_version5.hpp +++ b/trunk/src/core/srs_core_version5.hpp @@ -9,6 +9,6 @@ #define VERSION_MAJOR 5 #define VERSION_MINOR 0 -#define VERSION_REVISION 196 +#define VERSION_REVISION 197 #endif