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Fixed some big memory leaks with audio samples and ffmpeg (pprof ./openshot-example "/tmp/openshot-example.30691._main_-end.heap" --inuse_objects --lines --heapcheck --edgefraction=1e-10 --nodefraction=1e-10 --gv

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This commit is contained in:
Jonathan Thomas
2015-03-04 15:26:08 -06:00
parent c27e4a1426
commit b703f44f7c
4 changed files with 48 additions and 28 deletions
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41
src/FFmpegWriter.cpp
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@@ -1031,7 +1031,8 @@ void FFmpegWriter::write_audio_packets(bool final)
ChannelLayout channel_layout_in_frame = LAYOUT_MONO; // default channel layout
// Create a new array (to hold all S16 audio samples, for the current queued frames
int16_t* frame_samples = new int16_t[(queued_audio_frames.size() * AVCODEC_MAX_AUDIO_FRAME_SIZE) + FF_INPUT_BUFFER_PADDING_SIZE];
int16_t* queued_samples = new int16_t[(queued_audio_frames.size() * AVCODEC_MAX_AUDIO_FRAME_SIZE) + FF_INPUT_BUFFER_PADDING_SIZE];
int16_t* resampled_samples = NULL;
// Loop through each queued audio frame
while (!queued_audio_frames.empty())
@@ -1058,7 +1059,7 @@ void FFmpegWriter::write_audio_packets(bool final)
// Translate audio sample values back to 16 bit integers
for (int s = 0; s < total_frame_samples; s++, frame_position++)
// Translate sample value and copy into buffer
frame_samples[frame_position] = int(frame_samples_float[s] * (1 << 15));
queued_samples[frame_position] = int(frame_samples_float[s] * (1 << 15));
// Deallocate float array
@@ -1082,15 +1083,16 @@ void FFmpegWriter::write_audio_packets(bool final)
// Keep track of the original sample format
AVSampleFormat output_sample_fmt = audio_codec->sample_fmt;
AVFrame *audio_frame = NULL;
if (!final) {
// Create input frame (and allocate arrays)
AVFrame *audio_frame = avcodec_alloc_frame();
audio_frame = avcodec_alloc_frame();
avcodec_get_frame_defaults(audio_frame);
audio_frame->nb_samples = total_frame_samples / channels_in_frame;
av_samples_alloc(audio_frame->data, audio_frame->linesize, channels_in_frame, total_frame_samples / channels_in_frame, AV_SAMPLE_FMT_S16, 0);
//av_samples_alloc(audio_frame->data, audio_frame->linesize, channels_in_frame, total_frame_samples / channels_in_frame, AV_SAMPLE_FMT_S16, 0);
// Fill input frame with sample data
avcodec_fill_audio_frame(audio_frame, channels_in_frame, AV_SAMPLE_FMT_S16, (uint8_t *) frame_samples,
avcodec_fill_audio_frame(audio_frame, channels_in_frame, AV_SAMPLE_FMT_S16, (uint8_t *) queued_samples,
audio_frame->nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * channels_in_frame, 1);
// Do not convert audio to planar format (yet). We need to keep everything interleaved at this point.
@@ -1159,19 +1161,21 @@ void FFmpegWriter::write_audio_packets(bool final)
audio_frame->linesize[0], // input plane size, in bytes (0 if unknown)
audio_frame->nb_samples); // number of input samples to convert
// Create a new array (to hold all resampled S16 audio samples)
resampled_samples = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
// Copy audio samples over original samples
memcpy(frame_samples, audio_converted->data[0], nb_samples * av_get_bytes_per_sample(output_sample_fmt) * info.channels);
memcpy(resampled_samples, audio_converted->data[0], nb_samples * av_get_bytes_per_sample(output_sample_fmt) * info.channels);
// Update remaining samples (since we just resampled the audio, and things might have changed)
remaining_frame_samples = nb_samples * (float(av_get_bytes_per_sample(output_sample_fmt)) / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)) * info.channels;
#pragma omp critical (debug_output)
AppendDebugMethod("FFmpegWriter::write_audio_packets (Successfully completed 1st resampling)", "nb_samples", nb_samples, "remaining_frame_samples", remaining_frame_samples, "", -1, "", -1, "", -1, "", -1);
// Free AVFrames
avcodec_free_frame(&audio_frame); // TODO: Find a way to clear the memory inside this frame (memory leak)
// Remove converted audio
av_freep(&audio_converted[0]);
avcodec_free_frame(&audio_converted);
#pragma omp critical (debug_output)
AppendDebugMethod("FFmpegWriter::write_audio_packets (Successfully completed 1st resampling)", "nb_samples", nb_samples, "remaining_frame_samples", remaining_frame_samples, "", -1, "", -1, "", -1, "", -1);
}
// Loop until no more samples
@@ -1188,7 +1192,7 @@ void FFmpegWriter::write_audio_packets(bool final)
// Copy frame samples into the packet samples array
if (!final)
memcpy(samples + audio_input_position, frame_samples + samples_position, diff * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16));
memcpy(samples + audio_input_position, resampled_samples + samples_position, diff * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16));
// Increment counters
audio_input_position += diff;
@@ -1228,7 +1232,7 @@ void FFmpegWriter::write_audio_packets(bool final)
AVFrame *audio_frame = avcodec_alloc_frame();
avcodec_get_frame_defaults(audio_frame);
audio_frame->nb_samples = audio_input_position / info.channels;
av_samples_alloc(audio_frame->data, audio_frame->linesize, info.channels, audio_input_position / info.channels, output_sample_fmt, 0);
//av_samples_alloc(audio_frame->data, audio_frame->linesize, info.channels, audio_input_position / info.channels, output_sample_fmt, 0);
// Fill input frame with sample data
avcodec_fill_audio_frame(audio_frame, info.channels, output_sample_fmt, (uint8_t *) samples,
@@ -1339,8 +1343,15 @@ void FFmpegWriter::write_audio_packets(bool final)
final = false;
}
// Delete arrays
delete[] frame_samples;
// Free AVFrames
if (audio_frame) {
av_freep(&audio_frame[0]);
avcodec_free_frame(&audio_frame);
}
if (resampled_samples) {
delete[] resampled_samples;
resampled_samples = NULL;
}
} // end task
}