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Changing some libopenshot code based on Codacy feedback. Small refactoring, scope limiing.

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This commit is contained in:
Jonathan Thomas
2020-06-04 18:52:19 -05:00
parent 2834e7731b
commit 62d7fb5459
2 changed files with 13 additions and 31 deletions
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42
src/FFmpegWriter.cpp
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@@ -83,7 +83,7 @@ static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx, int6
}
#endif // HAVE_HW_ACCEL
FFmpegWriter::FFmpegWriter(std::string path) :
FFmpegWriter::FFmpegWriter(const std::string path) :
path(path), fmt(NULL), oc(NULL), audio_st(NULL), video_st(NULL), samples(NULL),
audio_outbuf(NULL), audio_outbuf_size(0), audio_input_frame_size(0), audio_input_position(0),
initial_audio_input_frame_size(0), img_convert_ctx(NULL), cache_size(8), num_of_rescalers(32),
@@ -864,9 +864,6 @@ void FFmpegWriter::flush_encoders() {
return;
#endif
int error_code = 0;
int stop_encoding = 1;
// FLUSH VIDEO ENCODER
if (info.has_video)
for (;;) {
@@ -940,13 +937,9 @@ void FFmpegWriter::flush_encoders() {
ZmqLogger::Instance()->AppendDebugMethod("FFmpegWriter::flush_encoders ERROR [" + (std::string) av_err2str(error_code) + "]", "error_code", error_code);
}
if (!got_packet) {
stop_encoding = 1;
break;
}
// Override PTS (in frames and scaled to the codec's timebase)
//pkt.pts = write_video_count;
// set the timestamp
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, video_codec->time_base, video_st->time_base);
@@ -961,10 +954,6 @@ void FFmpegWriter::flush_encoders() {
if (error_code < 0) {
ZmqLogger::Instance()->AppendDebugMethod("FFmpegWriter::flush_encoders ERROR [" + (std::string)av_err2str(error_code) + "]", "error_code", error_code);
}
// Deallocate memory (if needed)
if (video_outbuf)
av_freep(&video_outbuf);
}
// FLUSH AUDIO ENCODER
@@ -986,20 +975,15 @@ void FFmpegWriter::flush_encoders() {
pkt.pts = pkt.dts = write_audio_count;
/* encode the image */
int got_packet = 0;
int error_code = 0;
#if IS_FFMPEG_3_2
avcodec_send_frame(audio_codec, NULL);
got_packet = 0;
error_code = avcodec_send_frame(audio_codec, NULL);
#else
error_code = avcodec_encode_audio2(audio_codec, &pkt, NULL, &got_packet);
#endif
if (error_code < 0) {
ZmqLogger::Instance()->AppendDebugMethod("FFmpegWriter::flush_encoders ERROR [" + (std::string)av_err2str(error_code) + "]", "error_code", error_code);
}
if (!got_packet) {
stop_encoding = 1;
break;
}
// Since the PTS can change during encoding, set the value again. This seems like a huge hack,
// but it fixes lots of PTS related issues when I do this.
@@ -1583,29 +1567,27 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
channels_in_frame = frame->GetAudioChannelsCount();
channel_layout_in_frame = frame->ChannelsLayout();
// Get audio sample array
float *frame_samples_float = NULL;
// Get samples interleaved together (c1 c2 c1 c2 c1 c2)
frame_samples_float = frame->GetInterleavedAudioSamples(sample_rate_in_frame, NULL, &samples_in_frame);
// Calculate total samples
total_frame_samples = samples_in_frame * channels_in_frame;
// Translate audio sample values back to 16 bit integers with saturation
float valF;
int16_t conv;
const int16_t max16 = 32767;
const int16_t min16 = -32768;
for (int s = 0; s < total_frame_samples; s++, frame_position++) {
valF = frame_samples_float[s] * (1 << 15);
if (valF > max16)
float valF = frame_samples_float[s] * (1 << 15);
int16_t conv;
if (valF > max16) {
conv = max16;
else if (valF < min16)
} else if (valF < min16) {
conv = min16;
else
} else {
conv = int(valF + 32768.5) - 32768; // +0.5 is for rounding
}
// Copy into buffer
all_queued_samples[frame_position] = conv;
@@ -1731,10 +1713,11 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
// Determine how many samples we need
int diff = 0;
if (remaining_frame_samples >= remaining_packet_samples)
if (remaining_frame_samples >= remaining_packet_samples) {
diff = remaining_packet_samples;
else if (remaining_frame_samples < remaining_packet_samples)
} else {
diff = remaining_frame_samples;
}
// Copy frame samples into the packet samples array
if (!is_final)
@@ -1746,7 +1729,6 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
audio_input_position += diff;
samples_position += diff * (av_get_bytes_per_sample(output_sample_fmt) / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16));
remaining_frame_samples -= diff;
remaining_packet_samples -= diff;
// Do we have enough samples to proceed?
if (audio_input_position < (audio_input_frame_size * info.channels) && !is_final)