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From 71deaa48afab2bcf9df67b9b347b1f44aad3a9ce Mon Sep 17 00:00:00 2001
From: John Preston <johnprestonmail@gmail.com>
Date: Thu, 13 May 2021 15:17:54 +0400
Subject: [PATCH] Don't crash on voice recording error.
Fixes #16217.
---
.../media/audio/media_audio_capture.cpp | 89 +++++++++++--------
1 file changed, 54 insertions(+), 35 deletions(-)
diff --git a/Telegram/SourceFiles/media/audio/media_audio_capture.cpp b/Telegram/SourceFiles/media/audio/media_audio_capture.cpp
index a5965e0d1988..deb2474d7891 100644
--- a/Telegram/SourceFiles/media/audio/media_audio_capture.cpp
+++ b/Telegram/SourceFiles/media/audio/media_audio_capture.cpp
@@ -47,17 +47,17 @@ class Instance::Inner final : public QObject {
void start(Fn<void(Update)> updated, Fn<void()> error);
void stop(Fn<void(Result&&)> callback = nullptr);
- void timeout();
-
private:
- void processFrame(int32 offset, int32 framesize);
+ void process();
+
+ [[nodiscard]] bool processFrame(int32 offset, int32 framesize);
void fail();
- void writeFrame(AVFrame *frame);
+ [[nodiscard]] bool writeFrame(AVFrame *frame);
// Writes the packets till EAGAIN is got from av_receive_packet()
// Returns number of packets written or -1 on error
- int writePackets();
+ [[nodiscard]] int writePackets();
Fn<void(Update)> _updated;
Fn<void()> _error;
@@ -150,6 +150,7 @@ struct Instance::Inner::Private {
AVCodec *codec = nullptr;
AVCodecContext *codecContext = nullptr;
bool opened = false;
+ bool processing = false;
int srcSamples = 0;
int dstSamples = 0;
@@ -217,7 +218,7 @@ struct Instance::Inner::Private {
Instance::Inner::Inner(QThread *thread)
: d(std::make_unique<Private>())
-, _timer(thread, [=] { timeout(); }) {
+, _timer(thread, [=] { process(); }) {
moveToThread(thread);
}
@@ -226,10 +227,10 @@ Instance::Inner::~Inner() {
}
void Instance::Inner::fail() {
- Expects(_error != nullptr);
-
stop();
- _error();
+ if (const auto error = base::take(_error)) {
+ InvokeQueued(this, error);
+ }
}
void Instance::Inner::start(Fn<void(Update)> updated, Fn<void()> error) {
@@ -384,13 +385,21 @@ void Instance::Inner::stop(Fn<void(Result&&)> callback) {
}
_timer.cancel();
- if (d->device) {
+ const auto needResult = (callback != nullptr);
+ const auto hadDevice = (d->device != nullptr);
+ if (hadDevice) {
alcCaptureStop(d->device);
- timeout(); // get last data
+ if (d->processing) {
+ Assert(!needResult); // stop in the middle of processing - error.
+ } else {
+ process(); // get last data
+ }
+ alcCaptureCloseDevice(d->device);
+ d->device = nullptr;
}
// Write what is left
- if (!_captured.isEmpty()) {
+ if (needResult && !_captured.isEmpty()) {
auto fadeSamples = kCaptureFadeInDuration * kCaptureFrequency / 1000;
auto capturedSamples = static_cast<int>(_captured.size() / sizeof(short));
if ((_captured.size() % sizeof(short)) || (d->fullSamples + capturedSamples < kCaptureFrequency) || (capturedSamples < fadeSamples)) {
@@ -414,11 +423,13 @@ void Instance::Inner::stop(Fn<void(Result&&)> callback) {
int32 framesize = d->srcSamples * d->codecContext->channels * sizeof(short), encoded = 0;
while (_captured.size() >= encoded + framesize) {
- processFrame(encoded, framesize);
+ if (!processFrame(encoded, framesize)) {
+ break;
+ }
encoded += framesize;
}
- writeFrame(nullptr); // drain the codec
- if (encoded != _captured.size()) {
+ // Drain the codec.
+ if (!writeFrame(nullptr) || encoded != _captured.size()) {
d->fullSamples = 0;
d->dataPos = 0;
d->data.clear();
@@ -436,14 +447,14 @@ void Instance::Inner::stop(Fn<void(Result&&)> callback) {
_captured = QByteArray();
// Finish stream
- if (d->device) {
+ if (needResult && hadDevice) {
av_write_trailer(d->fmtContext);
}
QByteArray result = d->fullSamples ? d->data : QByteArray();
VoiceWaveform waveform;
qint32 samples = d->fullSamples;
- if (samples && !d->waveform.isEmpty()) {
+ if (needResult && samples && !d->waveform.isEmpty()) {
int64 count = d->waveform.size(), sum = 0;
if (count >= Player::kWaveformSamplesCount) {
QVector<uint16> peaks;
@@ -472,11 +483,7 @@ void Instance::Inner::stop(Fn<void(Result&&)> callback) {
}
}
}
- if (d->device) {
- alcCaptureStop(d->device);
- alcCaptureCloseDevice(d->device);
- d->device = nullptr;
-
+ if (hadDevice) {
if (d->codecContext) {
avcodec_free_context(&d->codecContext);
d->codecContext = nullptr;
@@ -528,12 +535,17 @@ void Instance::Inner::stop(Fn<void(Result&&)> callback) {
d->waveform.clear();
}
- if (callback) {
+ if (needResult) {
callback({ result, waveform, samples });
}
}
-void Instance::Inner::timeout() {
+void Instance::Inner::process() {
+ Expects(!d->processing);
+
+ d->processing = true;
+ const auto guard = gsl::finally([&] { d->processing = false; });
+
if (!d->device) {
_timer.cancel();
return;
@@ -582,7 +594,9 @@ void Instance::Inner::timeout() {
// Write frames
int32 framesize = d->srcSamples * d->codecContext->channels * sizeof(short), encoded = 0;
while (uint32(_captured.size()) >= encoded + framesize + fadeSamples * sizeof(short)) {
- processFrame(encoded, framesize);
+ if (!processFrame(encoded, framesize)) {
+ return;
+ }
encoded += framesize;
}
@@ -597,13 +611,13 @@ void Instance::Inner::timeout() {
}
}
-void Instance::Inner::processFrame(int32 offset, int32 framesize) {
+bool Instance::Inner::processFrame(int32 offset, int32 framesize) {
// Prepare audio frame
if (framesize % sizeof(short)) { // in the middle of a sample
LOG(("Audio Error: Bad framesize in writeFrame() for capture, framesize %1, %2").arg(framesize));
fail();
- return;
+ return false;
}
auto samplesCnt = static_cast<int>(framesize / sizeof(short));
@@ -650,7 +664,7 @@ void Instance::Inner::processFrame(int32 offset, int32 framesize) {
if ((res = av_samples_alloc(d->dstSamplesData, 0, d->codecContext->channels, d->dstSamples, d->codecContext->sample_fmt, 1)) < 0) {
LOG(("Audio Error: Unable to av_samples_alloc for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
fail();
- return;
+ return false;
}
d->dstSamplesSize = av_samples_get_buffer_size(0, d->codecContext->channels, d->maxDstSamples, d->codecContext->sample_fmt, 0);
}
@@ -658,7 +672,7 @@ void Instance::Inner::processFrame(int32 offset, int32 framesize) {
if ((res = swr_convert(d->swrContext, d->dstSamplesData, d->dstSamples, (const uint8_t **)srcSamplesData, d->srcSamples)) < 0) {
LOG(("Audio Error: Unable to swr_convert for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
fail();
- return;
+ return false;
}
// Write audio frame
@@ -670,45 +684,50 @@ void Instance::Inner::processFrame(int32 offset, int32 framesize) {
avcodec_fill_audio_frame(frame, d->codecContext->channels, d->codecContext->sample_fmt, d->dstSamplesData[0], d->dstSamplesSize, 0);
- writeFrame(frame);
+ if (!writeFrame(frame)) {
+ return false;
+ }
d->fullSamples += samplesCnt;
av_frame_free(&frame);
+ return true;
}
-void Instance::Inner::writeFrame(AVFrame *frame) {
+bool Instance::Inner::writeFrame(AVFrame *frame) {
int res = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
res = avcodec_send_frame(d->codecContext, frame);
if (res == AVERROR(EAGAIN)) {
- int packetsWritten = writePackets();
+ const auto packetsWritten = writePackets();
if (packetsWritten < 0) {
if (frame && packetsWritten == AVERROR_EOF) {
LOG(("Audio Error: EOF in packets received when EAGAIN was got in avcodec_send_frame()"));
fail();
}
- return;
+ return false;
} else if (!packetsWritten) {
LOG(("Audio Error: No packets received when EAGAIN was got in avcodec_send_frame()"));
fail();
- return;
+ return false;
}
res = avcodec_send_frame(d->codecContext, frame);
}
if (res < 0) {
LOG(("Audio Error: Unable to avcodec_send_frame for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
fail();
- return;
+ return false;
}
if (!frame) { // drain
if ((res = writePackets()) != AVERROR_EOF) {
LOG(("Audio Error: not EOF in packets received when draining the codec, result %1").arg(res));
fail();
+ return false;
}
}
+ return true;
}
int Instance::Inner::writePackets() {
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