Control: tags 1037842 + patch Control: tags 1037842 + pending
Dear maintainer, I've prepared an NMU for ring (versioned as 20230206.0~ds2-1.3) and uploaded it. Regards.
From: Gianfranco Costamagna <locutusofb...@debian.org>
To: 1037...@bugs.debian.org
Cc:
Bcc:
Subject: ring: diff for NMU version 20230206.0~ds2-1.3
Date: Sat, 02 Sep 2023 11:57:01 +0200
X-NMUDIFF-Version: 2.22.1ubuntu1
Control: tags 1037842 + patch
Control: tags 1037842 + pending
[Replace XX with correct value]
Dear maintainer,
I've prepared an NMU for ring (versioned as 20230206.0~ds2-1.3) and
uploaded it to DELAYED/XX. Please feel free to tell me if I
should delay it longer.
Regards.
diff -Nru ring-20230206.0~ds2/debian/changelog ring-20230206.0~ds2/debian/changelog
--- ring-20230206.0~ds2/debian/changelog 2023-07-01 23:23:01.000000000 +0200
+++ ring-20230206.0~ds2/debian/changelog 2023-09-02 10:00:03.000000000 +0200
@@ -1,3 +1,14 @@
+ring (20230206.0~ds2-1.3) unstable; urgency=medium
+
+ * Non-maintainer upload.
+ * debian/patches/d5e6509975bfaa67f7b3c54db72336b810454fc2.patch:
+ - gcc-13 upstream build fix (Closes: #1037842)
+ * debian/patches/bfa5a93114266caf791ff62c84918249c6d75bdf.patch:
+ * debian/patches/fc975f0cacde4b06c6adde3d7f0c02f71abfb38c.patch:
+ - ffmpeg 6.0 upstream build fix.
+
+ -- Gianfranco Costamagna <locutusofb...@debian.org> Sat, 02 Sep 2023 10:00:03 +0200
+
ring (20230206.0~ds2-1.2) unstable; urgency=medium
* Non-maintainer upload.
diff -Nru ring-20230206.0~ds2/debian/patches/bfa5a93114266caf791ff62c84918249c6d75bdf.patch ring-20230206.0~ds2/debian/patches/bfa5a93114266caf791ff62c84918249c6d75bdf.patch
--- ring-20230206.0~ds2/debian/patches/bfa5a93114266caf791ff62c84918249c6d75bdf.patch 1970-01-01 01:00:00.000000000 +0100
+++ ring-20230206.0~ds2/debian/patches/bfa5a93114266caf791ff62c84918249c6d75bdf.patch 2023-09-02 10:00:03.000000000 +0200
@@ -0,0 +1,256 @@
+From bfa5a93114266caf791ff62c84918249c6d75bdf Mon Sep 17 00:00:00 2001
+From: Adrien Beraud <adrien.ber...@savoirfairelinux.com>
+Date: Sun, 5 Mar 2023 11:05:25 -0500
+Subject: [PATCH] debug utils: use ffmpeg for WavWriter
+
+The previous writer was not working, the format was not
+recognized by most players.
+The new version uses ffmpeg APIs to produce working wav files
+(tested stereo s16 and mono f32).
+The API remains the same and it should be a drop-in
+replacement for existing code.
+
+Change-Id: I328e32427d94a44085ddb1567b5a1bbee5a6a8bc
+---
+ contrib/src/ffmpeg/rules.mak | 9 +-
+ src/debug_utils.h | 211 +++++++++++++++++------------------
+ 2 files changed, 111 insertions(+), 109 deletions(-)
+
+diff --git a/daemon/src/debug_utils.h b/daemon/src/debug_utils.h
+index 8572790599..3e519b7e7e 100644
+--- a/daemon/src/debug_utils.h
++++ b/daemon/src/debug_utils.h
+@@ -75,127 +75,122 @@ private:
+ };
+
+ /**
+- * Minimally invasive audio logger. Writes a wav file from raw PCM or AVFrame. Helps debug what goes
+- * wrong with audio.
++ * Audio logger. Writes a wav file from raw PCM or AVFrame. Helps debug what goes wrong with audio.
+ */
+-class WavWriter
+-{
++class WavWriter {
+ public:
+- WavWriter(std::string filename, AVFrame* frame)
+- : format_(static_cast<AVSampleFormat>(frame->format))
+- , channels_(frame->channels)
+- , planar_(av_sample_fmt_is_planar(format_))
+- , depth_(av_get_bytes_per_sample(format_))
+- , stepPerSample_(planar_ ? depth_ : depth_ * channels_)
++ WavWriter(const char* filename, AVFrame* frame)
+ {
+- std::vector<AVSampleFormat> v {AV_SAMPLE_FMT_FLT,
+- AV_SAMPLE_FMT_FLTP,
+- AV_SAMPLE_FMT_DBL,
+- AV_SAMPLE_FMT_DBLP};
+- f_ = std::ofstream(filename, std::ios_base::out | std::ios_base::binary);
+- f_.imbue(std::locale::classic());
+- f_ << "RIFF----WAVEfmt ";
+- if (std::find(v.begin(), v.end(), format_) == v.end()) {
+- write(16, 4); // Chunk size
+- write(1, 2); // WAVE_FORMAT_PCM
+- write(frame->channels, 2);
+- write(frame->sample_rate, 4);
+- write(frame->sample_rate * depth_ * frame->channels, 4); // Bytes per second
+- write(depth_ * frame->channels, 2); // Multi-channel sample size
+- write(8 * depth_, 2); // Bits per sample
+- f_ << "data";
+- dataChunk_ = f_.tellp();
+- f_ << "----";
+- } else {
+- write(18, 4); // Chunk size
+- write(3, 2); // Non PCM data
+- write(frame->channels, 2);
+- write(frame->sample_rate, 4);
+- write(frame->sample_rate * depth_ * frame->channels, 4); // Bytes per second
+- write(depth_ * frame->channels, 2); // Multi-channel sample size
+- write(8 * depth_, 2); // Bits per sample
+- write(0, 2); // Extension size
+- f_ << "fact";
+- write(4, 4); // Chunk size
+- factChunk_ = f_.tellp();
+- f_ << "----";
+- f_ << "data";
+- dataChunk_ = f_.tellp();
+- f_ << "----";
+- }
+- }
++ JAMI_WARNING("WavWriter(): {} ({}, {})", filename, av_get_sample_fmt_name((AVSampleFormat)frame->format), frame->sample_rate);
++ avformat_alloc_output_context2(&format_ctx_, nullptr, "wav", filename);
++ if (!format_ctx_)
++ throw std::runtime_error("Failed to allocate output format context");
+
+- ~WavWriter()
+- {
+- length_ = f_.tellp();
+- f_.seekp(dataChunk_);
+- write(length_ - dataChunk_ + 4, 4); // bytes_per_sample * channels * nb_samples
+- f_.seekp(4);
+- write(length_ - 8, 4);
+- if (factChunk_) {
+- f_.seekp(factChunk_);
+- write((length_ - dataChunk_ + 4) / depth_, 4); // channels * nb_samples
++ AVCodecID codec_id = AV_CODEC_ID_NONE;
++ switch (frame->format) {
++ case AV_SAMPLE_FMT_U8:
++ codec_id = AV_CODEC_ID_PCM_U8;
++ break;
++ case AV_SAMPLE_FMT_S16:
++ case AV_SAMPLE_FMT_S16P:
++ codec_id = AV_CODEC_ID_PCM_S16LE;
++ break;
++ case AV_SAMPLE_FMT_S32:
++ case AV_SAMPLE_FMT_S32P:
++ codec_id = AV_CODEC_ID_PCM_S32LE;
++ break;
++ case AV_SAMPLE_FMT_S64:
++ case AV_SAMPLE_FMT_S64P:
++ codec_id = AV_CODEC_ID_PCM_S64LE;
++ break;
++ case AV_SAMPLE_FMT_FLT:
++ case AV_SAMPLE_FMT_FLTP:
++ codec_id = AV_CODEC_ID_PCM_F32LE;
++ break;
++ case AV_SAMPLE_FMT_DBL:
++ codec_id = AV_CODEC_ID_PCM_F64LE;
++ break;
++ default:
++ throw std::runtime_error("Unsupported audio format");
+ }
+- f_.flush();
+- }
+
+- template<typename Word>
+- void write(Word value, unsigned size = sizeof(Word))
+- {
+- auto p = reinterpret_cast<unsigned char const*>(&value);
+- for (int i = 0; size; --size, ++i)
+- f_.put(p[i]);
++ auto codec = avcodec_find_encoder(codec_id);
++ if (!codec)
++ throw std::runtime_error("Failed to find audio codec");
++
++ codec_ctx_ = avcodec_alloc_context3(codec);
++ if (!codec_ctx_)
++ throw std::runtime_error("Failed to allocate audio codec context");
++
++ codec_ctx_->sample_fmt = (AVSampleFormat)frame->format;
++ codec_ctx_->channel_layout = frame->channel_layout;
++ codec_ctx_->sample_rate = frame->sample_rate;
++ codec_ctx_->channels = frame->channels;
++ if (format_ctx_->oformat->flags & AVFMT_GLOBALHEADER)
++ codec_ctx_->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
++
++ if (avcodec_open2(codec_ctx_, codec, nullptr) < 0)
++ throw std::runtime_error("Failed to open audio codec");
++
++ stream_ = avformat_new_stream(format_ctx_, codec);
++ if (!stream_)
++ throw std::runtime_error("Failed to create audio stream");
++
++ if (avcodec_parameters_from_context(stream_->codecpar, codec_ctx_) < 0)
++ throw std::runtime_error("Failed to copy codec parameters to stream");
++
++ if (!(format_ctx_->oformat->flags & AVFMT_NOFILE)) {
++ if (avio_open(&format_ctx_->pb, filename, AVIO_FLAG_WRITE) < 0) {
++ throw std::runtime_error("Failed to open output file for writing");
++ }
++ }
++ if (avformat_write_header(format_ctx_, nullptr) < 0)
++ throw std::runtime_error("Failed to write header to output file");
+ }
+
+- void write(AVFrame* frame)
+- {
+- for (int c = 0; c < frame->channels; ++c) {
+- int offset = planar_ ? 0 : depth_ * c;
+- for (int i = 0; i < frame->nb_samples; ++i) {
+- uint8_t* p = &frame->extended_data[planar_ ? c : 0][i + offset];
+- switch (format_) {
+- case AV_SAMPLE_FMT_U8:
+- case AV_SAMPLE_FMT_U8P:
+- write<uint8_t>(*(uint8_t*) p);
+- break;
+- case AV_SAMPLE_FMT_S16:
+- case AV_SAMPLE_FMT_S16P:
+- write<int16_t>(*(int16_t*) p);
+- break;
+- case AV_SAMPLE_FMT_S32:
+- case AV_SAMPLE_FMT_S32P:
+- write<int32_t>(*(int32_t*) p);
+- break;
+- case AV_SAMPLE_FMT_S64:
+- case AV_SAMPLE_FMT_S64P:
+- write<int64_t>(*(int64_t*) p);
+- break;
+- case AV_SAMPLE_FMT_FLT:
+- case AV_SAMPLE_FMT_FLTP:
+- write<float>(*(float*) p);
+- break;
+- case AV_SAMPLE_FMT_DBL:
+- case AV_SAMPLE_FMT_DBLP:
+- write<double>(*(double*) p);
+- break;
+- default:
+- break;
+- }
++ void write(AVFrame* frame) {
++ int ret = avcodec_send_frame(codec_ctx_, frame);
++ if (ret < 0)
++ JAMI_ERROR("Error sending a frame to the encoder");
++ while (ret >= 0) {
++ AVPacket *pkt = av_packet_alloc();
++ ret = avcodec_receive_packet(codec_ctx_, pkt);
++ if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
++ break;
++ else if (ret < 0) {
++ JAMI_ERROR("Error encoding a frame");
++ break;
+ }
++ pkt->stream_index = stream_->index;
++ pkt->pts = lastPts;
++ pkt->dts = lastPts;
++ lastPts += frame->nb_samples * (int64_t)stream_->time_base.den / (stream_->time_base.num * (int64_t)frame->sample_rate);
++ ret = av_write_frame(format_ctx_, pkt);
++ if (ret < 0) {
++ JAMI_ERROR("Error while writing output packet");
++ break;
++ }
++ av_packet_free(&pkt);
+ }
+- f_.flush();
+ }
+
++ ~WavWriter() {
++ if (codec_ctx_) {
++ avcodec_close(codec_ctx_);
++ avcodec_free_context(&codec_ctx_);
++ }
++ if (format_ctx_) {
++ av_write_trailer(format_ctx_);
++ if (!(format_ctx_->oformat->flags & AVFMT_NOFILE))
++ avio_closep(&format_ctx_->pb);
++ avformat_free_context(format_ctx_);
++ }
++ }
+ private:
+- std::ofstream f_;
+- size_t dataChunk_ {0};
+- size_t factChunk_ {0};
+- size_t length_ {0};
+- AVSampleFormat format_ {AV_SAMPLE_FMT_NONE};
+- size_t channels_ {0};
+- bool planar_ {false};
+- int depth_ {0};
+- int stepPerSample_ {0};
++ AVFormatContext* format_ctx_ {nullptr};
++ AVCodecContext* codec_ctx_ {nullptr};
++ AVStream* stream_ {nullptr};
++ int64_t lastPts {0};
+ };
+
+ /**
+--
+GitLab
+
diff -Nru ring-20230206.0~ds2/debian/patches/d5e6509975bfaa67f7b3c54db72336b810454fc2.patch ring-20230206.0~ds2/debian/patches/d5e6509975bfaa67f7b3c54db72336b810454fc2.patch
--- ring-20230206.0~ds2/debian/patches/d5e6509975bfaa67f7b3c54db72336b810454fc2.patch 1970-01-01 01:00:00.000000000 +0100
+++ ring-20230206.0~ds2/debian/patches/d5e6509975bfaa67f7b3c54db72336b810454fc2.patch 2023-09-02 10:00:03.000000000 +0200
@@ -0,0 +1,65 @@
+From d5e6509975bfaa67f7b3c54db72336b810454fc2 Mon Sep 17 00:00:00 2001
+From: =?UTF-8?q?S=C3=A9bastien=20Blin?=
+ <sebastien.b...@savoirfairelinux.com>
+Date: Wed, 26 Apr 2023 12:38:01 -0400
+Subject: [PATCH] misc: fix build on gcc13
+
+Change-Id: I2a42352cd560ea8f48d8f052e2819d121da1057c
+---
+ src/archiver.h | 1 +
+ src/jami/trace-tools.h | 1 +
+ src/scheduled_executor.h | 1 +
+ src/string_utils.h | 1 +
+ 4 files changed, 4 insertions(+)
+
+diff --git a/daemon/src/archiver.h b/daemon/src/archiver.h
+index cba983ec9d..ccf7faa88a 100644
+--- a/daemon/src/archiver.h
++++ b/daemon/src/archiver.h
+@@ -22,6 +22,7 @@
+
+ #include "noncopyable.h"
+
++#include <cstdint>
+ #include <string>
+ #include <vector>
+ #include <map>
+diff --git a/daemon/src/jami/trace-tools.h b/daemon/src/jami/trace-tools.h
+index 6155a46c41..ccd65cdf83 100644
+--- a/daemon/src/jami/trace-tools.h
++++ b/daemon/src/jami/trace-tools.h
+@@ -34,6 +34,7 @@
+
+ #ifdef HAVE_CXXABI_H
+ #include <cxxabi.h>
++#include <string>
+
+ template<typename T>
+ std::string demangle()
+diff --git a/daemon/src/scheduled_executor.h b/daemon/src/scheduled_executor.h
+index b1fd34f04d..4802d3bcf9 100644
+--- a/daemon/src/scheduled_executor.h
++++ b/daemon/src/scheduled_executor.h
+@@ -29,6 +29,7 @@
+ #include <mutex>
+ #include <condition_variable>
+ #include <ciso646>
++#include <string>
+
+ #include "noncopyable.h"
+
+diff --git a/daemon/src/string_utils.h b/daemon/src/string_utils.h
+index 695e431c36..04f3133ef1 100644
+--- a/daemon/src/string_utils.h
++++ b/daemon/src/string_utils.h
+@@ -21,6 +21,7 @@
+
+ #pragma once
+
++#include <cstdint>
+ #include <string>
+ #include <vector>
+ #include <set>
+--
+GitLab
+
diff -Nru ring-20230206.0~ds2/debian/patches/fc975f0cacde4b06c6adde3d7f0c02f71abfb38c.patch ring-20230206.0~ds2/debian/patches/fc975f0cacde4b06c6adde3d7f0c02f71abfb38c.patch
--- ring-20230206.0~ds2/debian/patches/fc975f0cacde4b06c6adde3d7f0c02f71abfb38c.patch 1970-01-01 01:00:00.000000000 +0100
+++ ring-20230206.0~ds2/debian/patches/fc975f0cacde4b06c6adde3d7f0c02f71abfb38c.patch 2023-09-02 10:00:03.000000000 +0200
@@ -0,0 +1,594 @@
+From fc975f0cacde4b06c6adde3d7f0c02f71abfb38c Mon Sep 17 00:00:00 2001
+From: Aline Gondim Santos <aline.gondimsan...@savoirfairelinux.com>
+Date: Wed, 1 Mar 2023 11:44:22 -0300
+Subject: [PATCH] contrib: ffmpeg 6.0
+
+Change-Id: Ib5ff688d4ad6127e97d15ddcdd28368a3eb5d5cc
+---
+ contrib/daemon/src/ffmpeg/SHA512SUMS | 2 +-
+ .../src/ffmpeg/libopusdec-enable-FEC.patch | 24 +++--
+ .../nvenc-fix-reorderqueueflush-crash.patch | 21 ++++
+ contrib/daemon/src/ffmpeg/package.json | 6 +-
+ contrib/daemon/src/ffmpeg/rules.mak | 3 +-
+ .../ffmpeg/windows-configure-ffnvcodec.patch | 8 +-
+ contrib/daemon/src/ffnvcodec/package.json | 2 +-
+ contrib/daemon/src/ffnvcodec/rules.mak | 2 +-
+ contrib/daemon/src/media-sdk/package.json | 6 +-
+ .../media-sdk/windows-static-lib-build.patch | 99 ++++++++++---------
+ contrib/daemon/src/x264/package.json | 2 +-
+ src/client/videomanager.cpp | 17 ++--
+ src/debug_utils.h | 3 +-
+ src/media/audio/audio_frame_resizer.cpp | 4 +-
+ src/media/audio/audiobuffer.cpp | 4 +-
+ src/media/audio/coreaudio/ios/corelayer.mm | 2 +-
+ src/media/audio/jack/jacklayer.cpp | 2 +-
+ src/media/audio/opensl/opensllayer.cpp | 4 +-
+ src/media/audio/portaudio/portaudiolayer.cpp | 2 +-
+ src/media/audio/resampler.cpp | 21 ++--
+ src/media/audio/resampler.h | 2 +-
+ src/media/libav_utils.cpp | 2 +-
+ src/media/media_decoder.cpp | 18 ++--
+ src/media/media_encoder.cpp | 9 +-
+ src/media/media_filter.cpp | 4 +-
+ src/media/media_recorder.cpp | 2 +-
+ src/media/media_stream.h | 4 +-
+ src/media/video/accel.cpp | 1 -
+ .../media/audio/test_audio_frame_resizer.cpp | 3 +-
+ test/unitTest/media/audio/test_resampler.cpp | 3 +-
+ test/unitTest/media/test_media_decoder.cpp | 2 +-
+ test/unitTest/media/test_media_encoder.cpp | 3 +-
+ test/unitTest/media/test_media_filter.cpp | 26 +++--
+ 33 files changed, 168 insertions(+), 145 deletions(-)
+ create mode 100644 contrib/daemon/src/ffmpeg/nvenc-fix-reorderqueueflush-crash.patch
+
+diff --git a/daemon/src/client/videomanager.cpp b/daemon/src/client/videomanager.cpp
+index 93ee6a051f..7fd72eeabb 100644
+--- a/daemon/src/client/videomanager.cpp
++++ b/daemon/src/client/videomanager.cpp
+@@ -111,8 +111,7 @@ void
+ AudioFrame::setFormat(const jami::AudioFormat& format)
+ {
+ auto d = pointer();
+- d->channels = format.nb_channels;
+- d->channel_layout = av_get_default_channel_layout(format.nb_channels);
++ av_channel_layout_default(&d->ch_layout, format.nb_channels);
+ d->sample_rate = format.sample_rate;
+ d->format = format.sampleFormat;
+ }
+@@ -121,7 +120,7 @@ jami::AudioFormat
+ AudioFrame::getFormat() const
+ {
+ return {(unsigned) frame_->sample_rate,
+- (unsigned) frame_->channels,
++ (unsigned) frame_->ch_layout.nb_channels,
+ (AVSampleFormat) frame_->format};
+ }
+
+@@ -149,7 +148,7 @@ AudioFrame::mix(const AudioFrame& frame)
+ {
+ auto& f = *pointer();
+ auto& fIn = *frame.pointer();
+- if (f.channels != fIn.channels || f.format != fIn.format || f.sample_rate != fIn.sample_rate) {
++ if (f.ch_layout.nb_channels != fIn.ch_layout.nb_channels || f.format != fIn.format || f.sample_rate != fIn.sample_rate) {
+ throw std::invalid_argument("Can't mix frames with different formats");
+ }
+ if (f.nb_samples == 0) {
+@@ -160,8 +159,8 @@ AudioFrame::mix(const AudioFrame& frame)
+ }
+ AVSampleFormat fmt = (AVSampleFormat) f.format;
+ bool isPlanar = av_sample_fmt_is_planar(fmt);
+- unsigned samplesPerChannel = isPlanar ? f.nb_samples : f.nb_samples * f.channels;
+- unsigned channels = isPlanar ? f.channels : 1;
++ unsigned samplesPerChannel = isPlanar ? f.nb_samples : f.nb_samples * f.ch_layout.nb_channels;
++ unsigned channels = isPlanar ? f.ch_layout.nb_channels : 1;
+ if (fmt == AV_SAMPLE_FMT_S16 || fmt == AV_SAMPLE_FMT_S16P) {
+ for (unsigned i = 0; i < channels; i++) {
+ auto c = (int16_t*) f.extended_data[i];
+@@ -192,8 +191,8 @@ AudioFrame::calcRMS() const
+ double rms = 0.0;
+ auto fmt = static_cast<AVSampleFormat>(frame_->format);
+ bool planar = av_sample_fmt_is_planar(fmt);
+- int perChannel = planar ? frame_->nb_samples : frame_->nb_samples * frame_->channels;
+- int channels = planar ? frame_->channels : 1;
++ int perChannel = planar ? frame_->nb_samples : frame_->nb_samples * frame_->ch_layout.nb_channels;
++ int channels = planar ? frame_->ch_layout.nb_channels : 1;
+ if (fmt == AV_SAMPLE_FMT_S16 || fmt == AV_SAMPLE_FMT_S16P) {
+ for (int c = 0; c < channels; ++c) {
+ auto buf = reinterpret_cast<int16_t*>(frame_->extended_data[c]);
+@@ -216,7 +215,7 @@ AudioFrame::calcRMS() const
+ return 0.0;
+ }
+ // divide by the number of multi-byte samples
+- return sqrt(rms / (frame_->nb_samples * frame_->channels));
++ return sqrt(rms / (frame_->nb_samples * frame_->ch_layout.nb_channels));
+ }
+
+ #ifdef ENABLE_VIDEO
+diff --git a/daemon/src/debug_utils.h b/daemon/src/debug_utils.h
+index 3e519b7e7e..1ddcfd8457 100644
+--- a/daemon/src/debug_utils.h
++++ b/daemon/src/debug_utils.h
+@@ -123,9 +123,8 @@ public:
+ throw std::runtime_error("Failed to allocate audio codec context");
+
+ codec_ctx_->sample_fmt = (AVSampleFormat)frame->format;
+- codec_ctx_->channel_layout = frame->channel_layout;
++ codec_ctx_->ch_layout = frame->ch_layout;
+ codec_ctx_->sample_rate = frame->sample_rate;
+- codec_ctx_->channels = frame->channels;
+ if (format_ctx_->oformat->flags & AVFMT_GLOBALHEADER)
+ codec_ctx_->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+
+diff --git a/daemon/src/media/audio/audio_frame_resizer.cpp b/daemon/src/media/audio/audio_frame_resizer.cpp
+index a6eac9864e..0fc4bad937 100644
+--- a/daemon/src/media/audio/audio_frame_resizer.cpp
++++ b/daemon/src/media/audio/audio_frame_resizer.cpp
+@@ -95,10 +95,10 @@ AudioFrameResizer::enqueue(std::shared_ptr<AudioFrame>&& frame)
+
+ int ret = 0;
+ auto f = frame->pointer();
+- AudioFormat format(f->sample_rate, f->channels, (AVSampleFormat) f->format);
++ AudioFormat format(f->sample_rate, f->ch_layout.nb_channels, (AVSampleFormat) f->format);
+ if (format != format_) {
+ JAMI_ERR() << "Expected " << format_ << ", but got "
+- << AudioFormat(f->sample_rate, f->channels, (AVSampleFormat) f->format);
++ << AudioFormat(f->sample_rate, f->ch_layout.nb_channels, (AVSampleFormat) f->format);
+ setFormat(format, frameSize_);
+ }
+
+diff --git a/daemon/src/media/audio/audiobuffer.cpp b/daemon/src/media/audio/audiobuffer.cpp
+index 9ef33c8f1e..d3481b2f1a 100644
+--- a/daemon/src/media/audio/audiobuffer.cpp
++++ b/daemon/src/media/audio/audiobuffer.cpp
+@@ -348,9 +348,9 @@ AudioBuffer::append(const AudioFrame& audioFrame)
+ {
+ auto frame = audioFrame.pointer();
+ // FIXME we assume frame is s16 interleaved
+- if (channels() != static_cast<unsigned>(frame->channels)
++ if (channels() != static_cast<unsigned>(frame->ch_layout.nb_channels)
+ || getSampleRate() != frame->sample_rate) {
+- auto newFormat = AudioFormat {(unsigned) frame->sample_rate, (unsigned) frame->channels};
++ auto newFormat = AudioFormat {(unsigned) frame->sample_rate, (unsigned) frame->ch_layout.nb_channels};
+ setFormat(newFormat);
+ }
+
+diff --git a/daemon/src/media/audio/coreaudio/ios/corelayer.mm b/daemon/src/media/audio/coreaudio/ios/corelayer.mm
+index d6006de701..b58b712c3e 100644
+--- a/daemon/src/media/audio/coreaudio/ios/corelayer.mm
++++ b/daemon/src/media/audio/coreaudio/ios/corelayer.mm
+@@ -401,7 +401,7 @@ CoreLayer::write(AudioUnitRenderActionFlags* ioActionFlags,
+
+ if (auto toPlay = getPlayback(currentOutFormat, inNumberFrames)) {
+ const auto& frame = *toPlay->pointer();
+- for (unsigned i = 0; i < frame.channels; ++i) {
++ for (unsigned i = 0; i < frame.ch_layout.nb_channels; ++i) {
+ std::copy_n((Float32*)frame.extended_data[i], inNumberFrames, (Float32*)ioData->mBuffers[i].mData);
+ }
+ } else {
+diff --git a/daemon/src/media/audio/jack/jacklayer.cpp b/daemon/src/media/audio/jack/jacklayer.cpp
+index a9d64a5172..a0cdfa5d4c 100644
+--- a/daemon/src/media/audio/jack/jacklayer.cpp
++++ b/daemon/src/media/audio/jack/jacklayer.cpp
+@@ -110,7 +110,7 @@ JackLayer::write(const AudioFrame& buffer)
+ {
+ auto num_samples = buffer.pointer()->nb_samples;
+ auto num_bytes = num_samples * sizeof(float);
+- auto channels = std::min<size_t>(out_ringbuffers_.size(), buffer.pointer()->channels);
++ auto channels = std::min<size_t>(out_ringbuffers_.size(), buffer.pointer()->ch_layout.nb_channels);
+ for (size_t i = 0; i < channels; ++i) {
+ jack_ringbuffer_write(out_ringbuffers_[i],
+ (const char*) buffer.pointer()->extended_data[i],
+diff --git a/daemon/src/media/audio/opensl/opensllayer.cpp b/daemon/src/media/audio/opensl/opensllayer.cpp
+index 0ee84fa464..e0d11b32b7 100644
+--- a/daemon/src/media/audio/opensl/opensllayer.cpp
++++ b/daemon/src/media/audio/opensl/opensllayer.cpp
+@@ -267,7 +267,7 @@ OpenSLLayer::engineServicePlay()
+ sample_buf* buf;
+ while (player_ and freePlayBufQueue_.front(&buf)) {
+ if (auto dat = getToPlay(hardwareFormat_, hardwareBuffSize_)) {
+- buf->size_ = dat->pointer()->nb_samples * dat->pointer()->channels
++ buf->size_ = dat->pointer()->nb_samples * dat->pointer()->ch_layout.nb_channels
+ * sizeof(AudioSample);
+ if (buf->size_ > buf->cap_) {
+ JAMI_ERR("buf->size_(%zu) > buf->cap_(%zu)", buf->size_, buf->cap_);
+@@ -300,7 +300,7 @@ OpenSLLayer::engineServiceRing()
+ sample_buf* buf;
+ while (ringtone_ and freeRingBufQueue_.front(&buf)) {
+ if (auto dat = getToRing(hardwareFormat_, hardwareBuffSize_)) {
+- buf->size_ = dat->pointer()->nb_samples * dat->pointer()->channels
++ buf->size_ = dat->pointer()->nb_samples * dat->pointer()->ch_layout.nb_channels
+ * sizeof(AudioSample);
+ if (buf->size_ > buf->cap_) {
+ JAMI_ERR("buf->size_(%zu) > buf->cap_(%zu)", buf->size_, buf->cap_);
+diff --git a/daemon/src/media/audio/portaudio/portaudiolayer.cpp b/daemon/src/media/audio/portaudio/portaudiolayer.cpp
+index 40c9e2cdc1..81f23bd37e 100644
+--- a/daemon/src/media/audio/portaudio/portaudiolayer.cpp
++++ b/daemon/src/media/audio/portaudio/portaudiolayer.cpp
+@@ -736,7 +736,7 @@ PortAudioLayer::PortAudioLayerImpl::paOutputCallback(PortAudioLayer& parent,
+ return paContinue;
+ }
+
+- auto nFrames = toPlay->pointer()->nb_samples * toPlay->pointer()->channels;
++ auto nFrames = toPlay->pointer()->nb_samples * toPlay->pointer()->ch_layout.nb_channels;
+ std::copy_n((AudioSample*) toPlay->pointer()->extended_data[0], nFrames, outputBuffer);
+ return paContinue;
+ }
+diff --git a/daemon/src/media/audio/resampler.cpp b/daemon/src/media/audio/resampler.cpp
+index bce9eaa84c..211fc135fa 100644
+--- a/daemon/src/media/audio/resampler.cpp
++++ b/daemon/src/media/audio/resampler.cpp
+@@ -50,13 +50,11 @@ Resampler::reinit(const AVFrame* in, const AVFrame* out)
+ throw std::bad_alloc();
+ }
+
+- av_opt_set_int(swrCtx, "ich", in->channels, 0);
+- av_opt_set_int(swrCtx, "icl", in->channel_layout, 0);
++ av_opt_set_chlayout(swrCtx, "ichl", &in->ch_layout, 0);
+ av_opt_set_int(swrCtx, "isr", in->sample_rate, 0);
+ av_opt_set_sample_fmt(swrCtx, "isf", static_cast<AVSampleFormat>(in->format), 0);
+
+- av_opt_set_int(swrCtx, "och", out->channels, 0);
+- av_opt_set_int(swrCtx, "ocl", out->channel_layout, 0);
++ av_opt_set_chlayout(swrCtx, "ochl", &out->ch_layout, 0);
+ av_opt_set_int(swrCtx, "osr", out->sample_rate, 0);
+ av_opt_set_sample_fmt(swrCtx, "osf", static_cast<AVSampleFormat>(out->format), 0);
+
+@@ -71,10 +69,10 @@ Resampler::reinit(const AVFrame* in, const AVFrame* out)
+ * LFE downmixing is optional, so any coefficient can be used, we use +6dB for mono and
+ * +0dB in each channel for stereo.
+ */
+- if (in->channel_layout == AV_CH_LAYOUT_5POINT1
+- || in->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) {
++ if (in->ch_layout.u.mask == AV_CH_LAYOUT_5POINT1
++ || in->ch_layout.u.mask == AV_CH_LAYOUT_5POINT1_BACK) {
+ // NOTE MSVC can't allocate dynamic size arrays on the stack
+- if (out->channels == 2) {
++ if (out->ch_layout.nb_channels == 2) {
+ double matrix[2][6];
+ // L = 1.0*FL + 0.707*FC + 0.707*BL + 1.0*LFE
+ matrix[0][0] = 1;
+@@ -152,8 +150,7 @@ Resampler::resample(const AudioBuffer& dataIn, AudioBuffer& dataOut)
+ AudioFrame resampled;
+ auto output = resampled.pointer();
+ output->sample_rate = dataOut.getSampleRate();
+- output->channel_layout = av_get_default_channel_layout(dataOut.channels());
+- output->channels = dataOut.channels();
++ av_channel_layout_default(&output->ch_layout, dataOut.channels());
+ output->format = AV_SAMPLE_FMT_S16;
+
+ if (resample(input, output) < 0)
+@@ -162,14 +159,14 @@ Resampler::resample(const AudioBuffer& dataIn, AudioBuffer& dataOut)
+ dataOut.resize(output->nb_samples);
+ dataOut.deinterleave(reinterpret_cast<const AudioSample*>(output->extended_data[0]),
+ output->nb_samples,
+- output->channels);
++ output->ch_layout.nb_channels);
+ }
+
+ std::unique_ptr<AudioFrame>
+ Resampler::resample(std::unique_ptr<AudioFrame>&& in, const AudioFormat& format)
+ {
+ if (in->pointer()->sample_rate == (int) format.sample_rate
+- && in->pointer()->channels == (int) format.nb_channels
++ && in->pointer()->ch_layout.nb_channels == (int) format.nb_channels
+ && (AVSampleFormat) in->pointer()->format == format.sampleFormat) {
+ return std::move(in);
+ }
+@@ -191,7 +188,7 @@ Resampler::resample(std::shared_ptr<AudioFrame>&& in, const AudioFormat& format)
+ }
+
+ if (inPtr->sample_rate == (int) format.sample_rate
+- && inPtr->channels == (int) format.nb_channels
++ && inPtr->ch_layout.nb_channels == (int) format.nb_channels
+ && (AVSampleFormat) inPtr->format == format.sampleFormat) {
+ return std::move(in);
+ }
+diff --git a/daemon/src/media/audio/resampler.h b/daemon/src/media/audio/resampler.h
+index 3a5bad17e3..4b798f4a2c 100644
+--- a/daemon/src/media/audio/resampler.h
++++ b/daemon/src/media/audio/resampler.h
+@@ -47,7 +47,7 @@ public:
+ *
+ * Resample from @input format to @output format.
+ *
+- * NOTE: sample_rate, channel_layout, and format should be set on @output
++ * NOTE: sample_rate, ch_layout, and format should be set on @output
+ */
+ int resample(const AVFrame* input, AVFrame* output);
+
+diff --git a/daemon/src/media/libav_utils.cpp b/daemon/src/media/libav_utils.cpp
+index 3e45b1cd0e..60cee22898 100644
+--- a/daemon/src/media/libav_utils.cpp
++++ b/daemon/src/media/libav_utils.cpp
+@@ -267,7 +267,7 @@ fillWithSilence(AVFrame* frame)
+ int ret = av_samples_set_silence(frame->extended_data,
+ 0,
+ frame->nb_samples,
+- frame->channels,
++ frame->ch_layout.nb_channels,
+ (AVSampleFormat) frame->format);
+ if (ret < 0)
+ JAMI_ERR() << "Failed to fill frame with silence";
+diff --git a/daemon/src/media/media_decoder.cpp b/daemon/src/media/media_decoder.cpp
+index 64ce3d1ad1..1bece6d987 100644
+--- a/daemon/src/media/media_decoder.cpp
++++ b/daemon/src/media/media_decoder.cpp
+@@ -595,8 +595,6 @@ MediaDecoder::prepareDecoderContext()
+ if (avStream_->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
+ if (decoderCtx_->framerate.num == 0 || decoderCtx_->framerate.den == 0)
+ decoderCtx_->framerate = inputParams_.framerate;
+- if (decoderCtx_->framerate.num == 0 || decoderCtx_->framerate.den == 0)
+- decoderCtx_->framerate = av_inv_q(decoderCtx_->time_base);
+ if (decoderCtx_->framerate.num == 0 || decoderCtx_->framerate.den == 0)
+ decoderCtx_->framerate = {30, 1};
+ }
+@@ -643,6 +641,15 @@ MediaDecoder::decode(AVPacket& packet)
+ #endif
+ auto frame = f->pointer();
+ ret = avcodec_receive_frame(decoderCtx_, frame);
++ // time_base is not set in AVCodecContext for decoding
++ // fail to set it causes pts to be incorrectly computed down in the function
++ if (inputDecoder_->type == AVMEDIA_TYPE_VIDEO) {
++ decoderCtx_->time_base.num = decoderCtx_->framerate.den;
++ decoderCtx_->time_base.den = decoderCtx_->framerate.num;
++ } else {
++ decoderCtx_->time_base.num = 1;
++ decoderCtx_->time_base.den = decoderCtx_->sample_rate;
++ }
+ frame->time_base = decoderCtx_->time_base;
+ if (resolutionChangedCallback_) {
+ if (decoderCtx_->width != width_ or decoderCtx_->height != height_) {
+@@ -663,11 +670,8 @@ MediaDecoder::decode(AVPacket& packet)
+ frameFinished = 1;
+
+ if (frameFinished) {
+- // channel layout is needed if frame will be resampled
+- if (!frame->channel_layout)
+- frame->channel_layout = av_get_default_channel_layout(frame->channels);
+-
+- frame->format = (AVPixelFormat) correctPixFmt(frame->format);
++ if (inputDecoder_->type == AVMEDIA_TYPE_VIDEO)
++ frame->format = (AVPixelFormat) correctPixFmt(frame->format);
+ auto packetTimestamp = frame->pts; // in stream time base
+ frame->pts = av_rescale_q_rnd(av_gettime() - startTime_,
+ {1, AV_TIME_BASE},
+diff --git a/daemon/src/media/media_encoder.cpp b/daemon/src/media/media_encoder.cpp
+index 1316ac442d..b767e2f123 100644
+--- a/daemon/src/media/media_encoder.cpp
++++ b/daemon/src/media/media_encoder.cpp
+@@ -641,14 +641,11 @@ MediaEncoder::prepareEncoderContext(const AVCodec* outputCodec, bool is_video)
+ encoderCtx->sample_rate = std::max(8000, audioOpts_.sampleRate);
+ encoderCtx->time_base = AVRational {1, encoderCtx->sample_rate};
+ if (audioOpts_.nbChannels > 2 || audioOpts_.nbChannels < 1) {
+- encoderCtx->channels = std::clamp(audioOpts_.nbChannels, 1, 2);
++ audioOpts_.nbChannels = std::clamp(audioOpts_.nbChannels, 1, 2);
+ JAMI_ERR() << "[" << encoderName
+- << "] Clamping invalid channel count: " << audioOpts_.nbChannels << " -> "
+- << encoderCtx->channels;
+- } else {
+- encoderCtx->channels = audioOpts_.nbChannels;
++ << "] Clamping invalid channel count: " << audioOpts_.nbChannels;
+ }
+- encoderCtx->channel_layout = av_get_default_channel_layout(encoderCtx->channels);
++ av_channel_layout_default(&encoderCtx->ch_layout, audioOpts_.nbChannels);
+ if (audioOpts_.frameSize) {
+ encoderCtx->frame_size = audioOpts_.frameSize;
+ JAMI_DBG() << "[" << encoderName << "] Frame size " << encoderCtx->frame_size;
+diff --git a/daemon/src/media/media_filter.cpp b/daemon/src/media/media_filter.cpp
+index cb5023732b..8ad3799593 100644
+--- a/daemon/src/media/media_filter.cpp
++++ b/daemon/src/media/media_filter.cpp
+@@ -171,7 +171,7 @@ MediaFilter::feedInput(AVFrame* frame, const std::string& inputName)
+ if (ms.format != frame->format
+ || (ms.isVideo && (ms.width != frame->width || ms.height != frame->height))
+ || (!ms.isVideo
+- && (ms.sampleRate != frame->sample_rate || ms.nbChannels != frame->channels))) {
++ && (ms.sampleRate != frame->sample_rate || ms.nbChannels != frame->ch_layout.nb_channels))) {
+ ms.update(frame);
+ if ((ret = reinitialize()) < 0)
+ return fail("Failed to reinitialize filter with new input parameters", ret);
+@@ -283,7 +283,7 @@ MediaFilter::initInputFilter(AVFilterInOut* in, const MediaStream& msp)
+ buffersrc = avfilter_get_by_name("buffer");
+ } else {
+ params->sample_rate = msp.sampleRate;
+- params->channel_layout = av_get_default_channel_layout(msp.nbChannels);
++ av_channel_layout_default(¶ms->ch_layout, msp.nbChannels);
+ buffersrc = avfilter_get_by_name("abuffer");
+ }
+
+diff --git a/daemon/src/media/media_recorder.cpp b/daemon/src/media/media_recorder.cpp
+index efa8b0d53e..58fec39906 100644
+--- a/daemon/src/media/media_recorder.cpp
++++ b/daemon/src/media/media_recorder.cpp
+@@ -576,7 +576,7 @@ std::string
+ MediaRecorder::buildAudioFilter(const std::vector<MediaStream>& peers,
+ const MediaStream& local) const
+ {
+- std::string baseFilter = "aresample=osr=48000:ocl=stereo:osf=s16";
++ std::string baseFilter = "aresample=osr=48000:ochl=stereo:osf=s16";
+ std::stringstream a;
+
+ switch (peers.size()) {
+diff --git a/daemon/src/media/media_stream.h b/daemon/src/media/media_stream.h
+index 1ede1c6026..34c0ef7578 100644
+--- a/daemon/src/media/media_stream.h
++++ b/daemon/src/media/media_stream.h
+@@ -112,7 +112,7 @@ struct MediaStream
+ format = c->sample_fmt;
+ isVideo = false;
+ sampleRate = c->sample_rate;
+- nbChannels = c->channels;
++ nbChannels = c->ch_layout.nb_channels;
+ frameSize = c->frame_size;
+ break;
+ default:
+@@ -144,7 +144,7 @@ struct MediaStream
+ height = f->height;
+ } else {
+ sampleRate = f->sample_rate;
+- nbChannels = f->channels;
++ nbChannels = f->ch_layout.nb_channels;
+ timeBase = rational<int>(1, f->sample_rate);
+ if (!frameSize)
+ frameSize = f->nb_samples;
+diff --git a/daemon/src/media/video/accel.cpp b/daemon/src/media/video/accel.cpp
+index 344bfbd008..bf206e39c4 100644
+--- a/daemon/src/media/video/accel.cpp
++++ b/daemon/src/media/video/accel.cpp
+@@ -313,7 +313,6 @@ HardwareAccel::setDetails(AVCodecContext* codecCtx)
+ if (type_ == CODEC_DECODER) {
+ codecCtx->hw_device_ctx = av_buffer_ref(deviceCtx_);
+ codecCtx->get_format = getFormatCb;
+- codecCtx->thread_safe_callbacks = 1;
+ } else if (type_ == CODEC_ENCODER) {
+ if (framesCtx_)
+ // encoder doesn't need a device context, only a frame context
+diff --git a/daemon/test/unitTest/media/audio/test_audio_frame_resizer.cpp b/daemon/test/unitTest/media/audio/test_audio_frame_resizer.cpp
+index 5e783f264d..7209a4b49b 100644
+--- a/daemon/test/unitTest/media/audio/test_audio_frame_resizer.cpp
++++ b/daemon/test/unitTest/media/audio/test_audio_frame_resizer.cpp
+@@ -74,8 +74,7 @@ AudioFrameResizerTest::getFrame(int n)
+ auto frame = std::make_shared<AudioFrame>();
+ frame->pointer()->format = format_.sampleFormat;
+ frame->pointer()->sample_rate = format_.sample_rate;
+- frame->pointer()->channels = format_.nb_channels;
+- frame->pointer()->channel_layout = av_get_default_channel_layout(format_.nb_channels);
++ av_channel_layout_default(&frame->pointer()->ch_layout, format_.nb_channels);
+ frame->pointer()->nb_samples = n;
+ CPPUNIT_ASSERT(av_frame_get_buffer(frame->pointer(), 0) >= 0);
+ return frame;
+diff --git a/daemon/test/unitTest/media/audio/test_resampler.cpp b/daemon/test/unitTest/media/audio/test_resampler.cpp
+index 3869ab3024..ad5e4d911d 100644
+--- a/daemon/test/unitTest/media/audio/test_resampler.cpp
++++ b/daemon/test/unitTest/media/audio/test_resampler.cpp
+@@ -98,8 +98,7 @@ ResamplerTest::testAudioFrame()
+ auto output = out.pointer();
+ output->format = AV_SAMPLE_FMT_FLT;
+ output->sample_rate = 48000;
+- output->channel_layout = AV_CH_LAYOUT_STEREO;
+- output->channels = 2;
++ av_channel_layout_from_mask(&output->ch_layout, AV_CH_LAYOUT_STEREO);
+
+ int ret = resampler_->resample(input->pointer(), output);
+ CPPUNIT_ASSERT_MESSAGE(libav_utils::getError(ret).c_str(), ret >= 0);
+diff --git a/daemon/test/unitTest/media/test_media_decoder.cpp b/daemon/test/unitTest/media/test_media_decoder.cpp
+index 0ce268d728..1cb2ed3481 100644
+--- a/daemon/test/unitTest/media/test_media_decoder.cpp
++++ b/daemon/test/unitTest/media/test_media_decoder.cpp
+@@ -81,7 +81,7 @@ MediaDecoderTest::testAudioFile()
+
+ decoder_.reset(new MediaDecoder([this](const std::shared_ptr<MediaFrame>&& f) mutable {
+ CPPUNIT_ASSERT(f->pointer()->sample_rate == decoder_->getStream().sampleRate);
+- CPPUNIT_ASSERT(f->pointer()->channels == decoder_->getStream().nbChannels);
++ CPPUNIT_ASSERT(f->pointer()->ch_layout.nb_channels == decoder_->getStream().nbChannels);
+ }));
+ DeviceParams dev;
+ dev.input = filename_;
+diff --git a/daemon/test/unitTest/media/test_media_encoder.cpp b/daemon/test/unitTest/media/test_media_encoder.cpp
+index b077bdfa67..483af9927c 100644
+--- a/daemon/test/unitTest/media/test_media_encoder.cpp
++++ b/daemon/test/unitTest/media/test_media_encoder.cpp
+@@ -115,8 +115,7 @@ getAudioFrame(int sampleRate, int nbSamples, int nbChannels)
+ return nullptr;
+
+ frame->format = AV_SAMPLE_FMT_S16;
+- frame->channels = nbChannels;
+- frame->channel_layout = av_get_default_channel_layout(nbChannels);
++ av_channel_layout_default(&frame->ch_layout, nbChannels);
+ frame->nb_samples = nbSamples;
+ frame->sample_rate = sampleRate;
+
+diff --git a/daemon/test/unitTest/media/test_media_filter.cpp b/daemon/test/unitTest/media/test_media_filter.cpp
+index 1b2767c395..8d09b2c660 100644
+--- a/daemon/test/unitTest/media/test_media_filter.cpp
++++ b/daemon/test/unitTest/media/test_media_filter.cpp
+@@ -116,13 +116,12 @@ static void
+ fillAudioFrameProps(AVFrame* frame, const MediaStream& ms)
+ {
+ frame->format = ms.format;
+- frame->channel_layout = av_get_default_channel_layout(ms.nbChannels);
++ av_channel_layout_default(&frame->ch_layout, ms.nbChannels);
+ frame->nb_samples = ms.frameSize;
+ frame->sample_rate = ms.sampleRate;
+- frame->channels = ms.nbChannels;
+ CPPUNIT_ASSERT(frame->format > AV_SAMPLE_FMT_NONE);
+ CPPUNIT_ASSERT(frame->nb_samples > 0);
+- CPPUNIT_ASSERT(frame->channel_layout != 0);
++ CPPUNIT_ASSERT(frame->ch_layout.u.mask != 0);
+ }
+
+ void
+@@ -132,7 +131,6 @@ MediaFilterTest::testAudioFilter()
+
+ // constants
+ const constexpr int nbSamples = 100;
+- const constexpr int64_t channelLayout = AV_CH_LAYOUT_STEREO;
+ const constexpr int sampleRate = 44100;
+ const constexpr enum AVSampleFormat format = AV_SAMPLE_FMT_S16;
+
+@@ -140,17 +138,16 @@ MediaFilterTest::testAudioFilter()
+ AudioFrame af;
+ auto frame = af.pointer();
+ frame->format = format;
+- frame->channel_layout = channelLayout;
++ av_channel_layout_from_mask(&frame->ch_layout, AV_CH_LAYOUT_STEREO);
+ frame->nb_samples = nbSamples;
+ frame->sample_rate = sampleRate;
+- frame->channels = av_get_channel_layout_nb_channels(channelLayout);
+
+ // construct the filter parameters
+- auto params = MediaStream("in1", format, rational<int>(1, sampleRate), sampleRate, frame->channels, nbSamples);
++ auto params = MediaStream("in1", format, rational<int>(1, sampleRate), sampleRate, frame->ch_layout.nb_channels, nbSamples);
+
+ // allocate and fill frame buffers
+ CPPUNIT_ASSERT(av_frame_get_buffer(frame, 0) >= 0);
+- fill_samples(reinterpret_cast<uint16_t*>(frame->data[0]), sampleRate, nbSamples, frame->channels, 440.0);
++ fill_samples(reinterpret_cast<uint16_t*>(frame->data[0]), sampleRate, nbSamples, frame->ch_layout.nb_channels, 440.0);
+
+ // prepare filter
+ std::vector<MediaStream> vec;
+@@ -190,17 +187,17 @@ MediaFilterTest::testAudioMixing()
+ fillAudioFrameProps(frame1, vec[0]);
+ frame1->pts = i * frame1->nb_samples;
+ CPPUNIT_ASSERT(av_frame_get_buffer(frame1, 0) >= 0);
+- fill_samples(reinterpret_cast<uint16_t*>(frame1->data[0]), frame1->sample_rate, frame1->nb_samples, frame1->channels, 440.0, t1);
++ fill_samples(reinterpret_cast<uint16_t*>(frame1->data[0]), frame1->sample_rate, frame1->nb_samples, frame1->ch_layout.nb_channels, 440.0, t1);
+
+ fillAudioFrameProps(frame2, vec[1]);
+ frame2->pts = i * frame2->nb_samples;
+ CPPUNIT_ASSERT(av_frame_get_buffer(frame2, 0) >= 0);
+- fill_samples(reinterpret_cast<uint16_t*>(frame2->data[0]), frame2->sample_rate, frame2->nb_samples, frame2->channels, 329.6276, t2);
++ fill_samples(reinterpret_cast<uint16_t*>(frame2->data[0]), frame2->sample_rate, frame2->nb_samples, frame2->ch_layout.nb_channels, 329.6276, t2);
+
+ fillAudioFrameProps(frame3, vec[2]);
+ frame3->pts = i * frame3->nb_samples;
+ CPPUNIT_ASSERT(av_frame_get_buffer(frame3, 0) >= 0);
+- fill_samples(reinterpret_cast<uint16_t*>(frame3->data[0]), frame3->sample_rate, frame3->nb_samples, frame3->channels, 349.2282, t3);
++ fill_samples(reinterpret_cast<uint16_t*>(frame3->data[0]), frame3->sample_rate, frame3->nb_samples, frame3->ch_layout.nb_channels, 349.2282, t3);
+
+ // apply filter
+ CPPUNIT_ASSERT(filter_->feedInput(frame1, "a1") >= 0);
+@@ -318,17 +315,16 @@ MediaFilterTest::testReinit()
+ AudioFrame af;
+ auto frame = af.pointer();
+ frame->format = AV_SAMPLE_FMT_S16;
+- frame->channel_layout = AV_CH_LAYOUT_STEREO;
++ av_channel_layout_from_mask(&frame->ch_layout, AV_CH_LAYOUT_STEREO);
+ frame->nb_samples = 100;
+ frame->sample_rate = 44100;
+- frame->channels = 2;
+
+ // construct the filter parameters with different sample rate
+- auto params = MediaStream("in1", frame->format, rational<int>(1, 16000), 16000, frame->channels, frame->nb_samples);
++ auto params = MediaStream("in1", frame->format, rational<int>(1, 16000), 16000, frame->ch_layout.nb_channels, frame->nb_samples);
+
+ // allocate and fill frame buffers
+ CPPUNIT_ASSERT(av_frame_get_buffer(frame, 0) >= 0);
+- fill_samples(reinterpret_cast<uint16_t*>(frame->data[0]), frame->sample_rate, frame->nb_samples, frame->channels, 440.0);
++ fill_samples(reinterpret_cast<uint16_t*>(frame->data[0]), frame->sample_rate, frame->nb_samples, frame->ch_layout.nb_channels, 440.0);
+
+ // prepare filter
+ std::vector<MediaStream> vec;
+--
+GitLab
+
diff -Nru ring-20230206.0~ds2/debian/patches/series ring-20230206.0~ds2/debian/patches/series
--- ring-20230206.0~ds2/debian/patches/series 2023-02-26 16:37:15.000000000 +0100
+++ ring-20230206.0~ds2/debian/patches/series 2023-09-02 10:00:03.000000000 +0200
@@ -3,3 +3,6 @@
2000-jsoncpp-rename.patch
2010-dont-build-gnutls.patch
2020-dont-build-msgpack.patch
+d5e6509975bfaa67f7b3c54db72336b810454fc2.patch
+bfa5a93114266caf791ff62c84918249c6d75bdf.patch
+fc975f0cacde4b06c6adde3d7f0c02f71abfb38c.patch
OpenPGP_signature
Description: OpenPGP digital signature
