Fix memory leak in audio encoder

2 objects were not deleted.
* resampler leaked anyway
* codec context might be deleted by ffmpeg,
  the documentation recommends to delete it.

Since the variable was touched, it was renamed to proper name.

Change-Id: I94965e83b060b8db251b13c8c0d923cddf804741
Reviewed-by: Artem Dyomin <artem.dyomin@qt.io>
Reviewed-by: Jøger Hansegård <joger.hansegard@qt.io>
Reviewed-by: Pavel Dubsky <pavel.dubsky@qt.io>
(cherry picked from commit 9cda4c7a5906630d4a3306dae4cefd8331751685)
Reviewed-by: Qt Cherry-pick Bot <cherrypick_bot@qt-project.org>

5 files changed, 51 insertions, 43 deletions

diff --git a/src/plugins/multimedia/ffmpeg/qffmpeg_p.h b/src/plugins/multimedia/ffmpeg/qffmpeg_p.h
index 9c42f5693..2dc97c7e1 100644
--- a/src/plugins/multimedia/ffmpeg/qffmpeg_p.h
+++ b/src/plugins/multimedia/ffmpeg/qffmpeg_p.h
@@ -135,6 +135,8 @@ using AVHWFramesConstraintsUPtr = std::unique_ptr<
         AVHWFramesConstraints,
         AVDeleter<decltype(&av_hwframe_constraints_free), &av_hwframe_constraints_free>>;
 
+using SwrContextUPtr = std::unique_ptr<SwrContext, AVDeleter<decltype(&swr_free), &swr_free>>;
+
 using PixelOrSampleFormat = int;
 using AVScore = int;
 constexpr AVScore BestAVScore = std::numeric_limits<AVScore>::max();
diff --git a/src/plugins/multimedia/ffmpeg/qffmpegencoder.cpp b/src/plugins/multimedia/ffmpeg/qffmpegencoder.cpp
index cd0e040bb..a9579c417 100644
--- a/src/plugins/multimedia/ffmpeg/qffmpegencoder.cpp
+++ b/src/plugins/multimedia/ffmpeg/qffmpegencoder.cpp
@@ -322,7 +322,7 @@ void AudioEncoder::open()
 {
     AVSampleFormat requested = QFFmpegMediaFormatInfo::avSampleFormat(format.sampleFormat());
 
-    codec = avcodec_alloc_context3(avCodec);
+    codecContext.reset(avcodec_alloc_context3(avCodec));
 
     if (stream->time_base.num != 1 || stream->time_base.den != format.sampleRate()) {
         qCDebug(qLcFFmpegEncoder) << "Most likely, av_format_write_header changed time base from"
@@ -330,38 +330,43 @@ void AudioEncoder::open()
                                   << stream->time_base.num << "/" << stream->time_base.den;
     }
 
-    codec->time_base = stream->time_base;
+    codecContext->time_base = stream->time_base;
 
-    avcodec_parameters_to_context(codec, stream->codecpar);
+    avcodec_parameters_to_context(codecContext.get(), stream->codecpar);
 
     AVDictionaryHolder opts;
-    applyAudioEncoderOptions(settings, avCodec->name, codec, opts);
+    applyAudioEncoderOptions(settings, avCodec->name, codecContext.get(), opts);
 
-    int res = avcodec_open2(codec, avCodec, opts);
+    int res = avcodec_open2(codecContext.get(), avCodec, opts);
     qCDebug(qLcFFmpegEncoder) << "audio codec opened" << res;
-    qCDebug(qLcFFmpegEncoder) << "audio codec params: fmt=" << codec->sample_fmt << "rate=" << codec->sample_rate;
+    qCDebug(qLcFFmpegEncoder) << "audio codec params: fmt=" << codecContext->sample_fmt
+                              << "rate=" << codecContext->sample_rate;
 
-    if (codec->sample_fmt != requested) {
+    if (codecContext->sample_fmt != requested) {
+        SwrContext *resampler = nullptr;
 #if QT_FFMPEG_OLD_CHANNEL_LAYOUT
-        resampler = swr_alloc_set_opts(nullptr,  // we're allocating a new context
-                                       codec->channel_layout,  // out_ch_layout
-                                       codec->sample_fmt,    // out_sample_fmt
-                                       codec->sample_rate,                // out_sample_rate
-                                       av_get_default_channel_layout(format.channelCount()), // in_ch_layout
-                                       requested,   // in_sample_fmt
-                                       format.sampleRate(),                // in_sample_rate
-                                       0,                    // log_offset
-                                       nullptr);
+        resampler = swr_alloc_set_opts(
+                nullptr, // we're allocating a new context
+                codecContext->channel_layout, // out_ch_layout
+                codecContext->sample_fmt, // out_sample_fmt
+                codecContext->sample_rate, // out_sample_rate
+                av_get_default_channel_layout(format.channelCount()), // in_ch_layout
+                requested, // in_sample_fmt
+                format.sampleRate(), // in_sample_rate
+                0, // log_offset
+                nullptr);
 #else
         AVChannelLayout in_ch_layout = {};
         av_channel_layout_default(&in_ch_layout, format.channelCount());
-        swr_alloc_set_opts2(&resampler,  // we're allocating a new context
-                            &codec->ch_layout, codec->sample_fmt, codec->sample_rate,
-                            &in_ch_layout, requested, format.sampleRate(),
-                            0, nullptr);
+        swr_alloc_set_opts2(&resampler, // we're allocating a new context
+                            &codecContext->ch_layout, codecContext->sample_fmt,
+                            codecContext->sample_rate, &in_ch_layout, requested,
+                            format.sampleRate(), 0, nullptr);
 #endif
 
         swr_init(resampler);
+
+        this->resampler.reset(resampler);
     }
 }
 
@@ -385,7 +390,7 @@ void AudioEncoder::init()
 {
     open();
     if (input) {
-        input->setFrameSize(codec->frame_size);
+        input->setFrameSize(codecContext->frame_size);
     }
     qCDebug(qLcFFmpegEncoder) << "AudioEncoder::init started audio device thread.";
 }
@@ -394,7 +399,7 @@ void AudioEncoder::cleanup()
 {
     while (!audioBufferQueue.empty())
         loop();
-    while (avcodec_send_frame(codec, nullptr) == AVERROR(EAGAIN))
+    while (avcodec_send_frame(codecContext.get(), nullptr) == AVERROR(EAGAIN))
         retrievePackets();
     retrievePackets();
 }
@@ -409,7 +414,7 @@ void AudioEncoder::retrievePackets()
 {
     while (1) {
         AVPacketUPtr packet(av_packet_alloc());
-        int ret = avcodec_receive_packet(codec, packet.get());
+        int ret = avcodec_receive_packet(codecContext.get(), packet.get());
         if (ret < 0) {
             if (ret != AVERROR(EOF))
                 break;
@@ -437,28 +442,29 @@ void AudioEncoder::loop()
     retrievePackets();
 
     auto frame = makeAVFrame();
-    frame->format = codec->sample_fmt;
+    frame->format = codecContext->sample_fmt;
 #if QT_FFMPEG_OLD_CHANNEL_LAYOUT
-    frame->channel_layout = codec->channel_layout;
-    frame->channels = codec->channels;
+    frame->channel_layout = codecContext->channel_layout;
+    frame->channels = codecContext->channels;
 #else
-    frame->ch_layout = codec->ch_layout;
+    frame->ch_layout = codecContext->ch_layout;
 #endif
-    frame->sample_rate = codec->sample_rate;
+    frame->sample_rate = codecContext->sample_rate;
     frame->nb_samples = buffer.frameCount();
     if (frame->nb_samples)
         av_frame_get_buffer(frame.get(), 0);
 
     if (resampler) {
         const uint8_t *data = buffer.constData<uint8_t>();
-        swr_convert(resampler, frame->extended_data, frame->nb_samples, &data, frame->nb_samples);
+        swr_convert(resampler.get(), frame->extended_data, frame->nb_samples, &data,
+                    frame->nb_samples);
     } else {
         memcpy(frame->buf[0]->data, buffer.constData<uint8_t>(), buffer.byteCount());
     }
 
     const auto &timeBase = stream->time_base;
     const auto pts = timeBase.den && timeBase.num
-            ? timeBase.den * samplesWritten / (codec->sample_rate * timeBase.num)
+            ? timeBase.den * samplesWritten / (codecContext->sample_rate * timeBase.num)
             : samplesWritten;
     setAVFrameTime(*frame, pts, timeBase);
     samplesWritten += buffer.frameCount();
@@ -469,7 +475,7 @@ void AudioEncoder::loop()
     //    qCDebug(qLcFFmpegEncoder) << "sending audio frame" << buffer.byteCount() << frame->pts <<
     //    ((double)buffer.frameCount()/frame->sample_rate);
 
-    int ret = avcodec_send_frame(codec, frame.get());
+    int ret = avcodec_send_frame(codecContext.get(), frame.get());
     if (ret < 0) {
         char errStr[1024];
         av_strerror(ret, errStr, 1024);
diff --git a/src/plugins/multimedia/ffmpeg/qffmpegencoder_p.h b/src/plugins/multimedia/ffmpeg/qffmpegencoder_p.h
index a2b64c09a..3e6306411 100644
--- a/src/plugins/multimedia/ffmpeg/qffmpegencoder_p.h
+++ b/src/plugins/multimedia/ffmpeg/qffmpegencoder_p.h
@@ -158,11 +158,11 @@ private:
     void loop() override;
 
     AVStream *stream = nullptr;
-    AVCodecContext *codec = nullptr;
-    QFFmpegAudioInput *input;
+    AVCodecContextUPtr codecContext;
+    QFFmpegAudioInput *input = nullptr;
     QAudioFormat format;
 
-    SwrContext *resampler = nullptr;
+    SwrContextUPtr resampler;
     qint64 samplesWritten = 0;
     const AVCodec *avCodec = nullptr;
     QMediaEncoderSettings settings;
diff --git a/src/plugins/multimedia/ffmpeg/qffmpegresampler.cpp b/src/plugins/multimedia/ffmpeg/qffmpegresampler.cpp
index 8b3c23f6c..f84aecfbc 100644
--- a/src/plugins/multimedia/ffmpeg/qffmpegresampler.cpp
+++ b/src/plugins/multimedia/ffmpeg/qffmpegresampler.cpp
@@ -33,6 +33,7 @@ Resampler::Resampler(const Codec *codec, const QAudioFormat &outputFormat)
 
 
     qCDebug(qLcResampler) << "init resampler" << m_outputFormat.sampleRate() << config << codecpar->sample_rate;
+    SwrContext *resampler = nullptr;
 #if QT_FFMPEG_OLD_CHANNEL_LAYOUT
     auto inConfig = codecpar->channel_layout;
     if (inConfig == 0)
@@ -61,12 +62,10 @@ Resampler::Resampler(const Codec *codec, const QAudioFormat &outputFormat)
                         nullptr);
 #endif
     swr_init(resampler);
+    m_resampler.reset(resampler);
 }
 
-Resampler::~Resampler()
-{
-    swr_free(&resampler);
-}
+Resampler::~Resampler() = default;
 
 QAudioBuffer Resampler::resample(const AVFrame *frame)
 {
@@ -75,7 +74,8 @@ QAudioBuffer Resampler::resample(const AVFrame *frame)
     QByteArray samples(m_outputFormat.bytesForFrames(maxOutSamples), Qt::Uninitialized);
     auto **in = const_cast<const uint8_t **>(frame->extended_data);
     auto *out = reinterpret_cast<uint8_t *>(samples.data());
-    const int outSamples = swr_convert(resampler, &out, maxOutSamples, in, frame->nb_samples);
+    const int outSamples =
+            swr_convert(m_resampler.get(), &out, maxOutSamples, in, frame->nb_samples);
 
     samples.resize(m_outputFormat.bytesForFrames(outSamples));
 
@@ -89,7 +89,7 @@ QAudioBuffer Resampler::resample(const AVFrame *frame)
 
 int Resampler::adjustMaxOutSamples(const AVFrame *frame)
 {
-    int maxOutSamples = swr_get_out_samples(resampler, frame->nb_samples);
+    int maxOutSamples = swr_get_out_samples(m_resampler.get(), frame->nb_samples);
 
     const auto remainingCompensationDistance = m_endCompensationSample - m_samplesProcessed;
 
@@ -101,7 +101,7 @@ int Resampler::adjustMaxOutSamples(const AVFrame *frame)
         // however it's not significant for our logic, in fact.
         // TODO: probably, it will need some improvements
         setSampleCompensation(0, 0);
-        maxOutSamples = swr_get_out_samples(resampler, frame->nb_samples);
+        maxOutSamples = swr_get_out_samples(m_resampler.get(), frame->nb_samples);
     }
 
     return maxOutSamples;
@@ -109,7 +109,7 @@ int Resampler::adjustMaxOutSamples(const AVFrame *frame)
 
 void Resampler::setSampleCompensation(qint32 delta, quint32 distance)
 {
-    const int res = swr_set_compensation(resampler, delta, static_cast<int>(distance));
+    const int res = swr_set_compensation(m_resampler.get(), delta, static_cast<int>(distance));
     if (res < 0)
         qCWarning(qLcResampler) << "swr_set_compensation fail:" << res;
     else {
diff --git a/src/plugins/multimedia/ffmpeg/qffmpegresampler_p.h b/src/plugins/multimedia/ffmpeg/qffmpegresampler_p.h
index 3e67103b4..55802a17d 100644
--- a/src/plugins/multimedia/ffmpeg/qffmpegresampler_p.h
+++ b/src/plugins/multimedia/ffmpeg/qffmpegresampler_p.h
@@ -40,7 +40,7 @@ private:
 
 private:
     QAudioFormat m_outputFormat;
-    SwrContext *resampler = nullptr;
+    SwrContextUPtr m_resampler;
     qint64 m_samplesProcessed = 0;
     qint64 m_endCompensationSample = std::numeric_limits<qint64>::min();
     qint32 m_sampleCompensationDelta = 0;