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FFmpegWriter: Code reformatting

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This commit is contained in:
FeRD (Frank Dana)
2020-03-05 01:38:25 -05:00
parent 3a88d21866
commit 92ebd15545
1 changed files with 75 additions and 34 deletions
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109
src/FFmpegWriter.cpp
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@@ -255,10 +255,14 @@ void FFmpegWriter::SetVideoOptions(bool has_video, std::string codec, Fraction f
info.pixel_ratio.num = pixel_ratio.num;
info.pixel_ratio.den = pixel_ratio.den;
}
if (bit_rate >= 1000) // bit_rate is the bitrate in b/s
if (bit_rate >= 1000) {
// bit_rate is the bitrate in b/s
info.video_bit_rate = bit_rate;
if ((bit_rate >= 0) && (bit_rate < 64)) // bit_rate is the bitrate in crf
}
if ((bit_rate >= 0) && (bit_rate < 64)) {
// bit_rate is the bitrate in crf
info.video_bit_rate = bit_rate;
}
info.interlaced_frame = interlaced;
info.top_field_first = top_field_first;
@@ -282,8 +286,10 @@ void FFmpegWriter::SetVideoOptions(bool has_video, std::string codec, Fraction f
// Set video export options (overloaded function)
void FFmpegWriter::SetVideoOptions(std::string codec, int width, int height, Fraction fps, int bit_rate) {
// Call full signature with some default parameters
FFmpegWriter::SetVideoOptions(true, codec, fps, width, height,
openshot::Fraction(1, 1), false, true, bit_rate);
FFmpegWriter::SetVideoOptions(
true, codec, fps, width, height,
openshot::Fraction(1, 1), false, true, bit_rate
);
}
@@ -326,8 +332,10 @@ void FFmpegWriter::SetAudioOptions(bool has_audio, std::string codec, int sample
// Set audio export options (overloaded function)
void FFmpegWriter::SetAudioOptions(std::string codec, int sample_rate, int bit_rate) {
// Call full signature with some default parameters
FFmpegWriter::SetAudioOptions(true, codec, sample_rate, 2,
openshot::LAYOUT_STEREO, bit_rate);
FFmpegWriter::SetAudioOptions(
true, codec, sample_rate, 2,
openshot::LAYOUT_STEREO, bit_rate
);
}
@@ -418,6 +426,7 @@ void FFmpegWriter::SetOption(StreamType stream, std::string name, std::string va
{
switch (c->codec_id) {
#if (LIBAVCODEC_VERSION_MAJOR >= 58)
// FFmpeg 4.0+
case AV_CODEC_ID_AV1 :
c->bit_rate = 0;
av_opt_set_int(c->priv_data, "qp", std::min(std::stoi(value),63), 0); // 0-63
@@ -475,6 +484,7 @@ void FFmpegWriter::SetOption(StreamType stream, std::string name, std::string va
{
switch (c->codec_id) {
#if (LIBAVCODEC_VERSION_MAJOR >= 58)
// FFmpeg 4.0+
case AV_CODEC_ID_AV1 :
c->bit_rate = 0;
av_opt_set_int(c->priv_data, "crf", std::min(std::stoi(value),63), 0);
@@ -1570,20 +1580,25 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
int nb_samples = 0;
// Convert audio samples
nb_samples = SWR_CONVERT(avr, // audio resample context
audio_converted->data, // output data pointers
audio_converted->linesize[0], // output plane size, in bytes. (0 if unknown)
audio_converted->nb_samples, // maximum number of samples that the output buffer can hold
audio_frame->data, // input data pointers
audio_frame->linesize[0], // input plane size, in bytes (0 if unknown)
audio_frame->nb_samples); // number of input samples to convert
nb_samples = SWR_CONVERT(
avr, // audio resample context
audio_converted->data, // output data pointers
audio_converted->linesize[0], // output plane size, in bytes. (0 if unknown)
audio_converted->nb_samples, // maximum number of samples that the output buffer can hold
audio_frame->data, // input data pointers
audio_frame->linesize[0], // input plane size, in bytes (0 if unknown)
audio_frame->nb_samples // number of input samples to convert
);
// Set remaining samples
remaining_frame_samples = nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
// Create a new array (to hold all resampled S16 audio samples)
all_resampled_samples = (int16_t *) av_malloc(
sizeof(int16_t) * nb_samples * info.channels * (av_get_bytes_per_sample(output_sample_fmt) / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)));
sizeof(int16_t) * nb_samples * info.channels
* (av_get_bytes_per_sample(output_sample_fmt) /
av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) )
);
// Copy audio samples over original samples
memcpy(all_resampled_samples, audio_converted->data[0], nb_samples * info.channels * av_get_bytes_per_sample(output_sample_fmt));
@@ -1613,8 +1628,14 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
// Copy frame samples into the packet samples array
if (!is_final)
//TODO: Make this more sane
memcpy(samples + (audio_input_position * (av_get_bytes_per_sample(output_sample_fmt) / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16))),
all_resampled_samples + samples_position, diff * av_get_bytes_per_sample(output_sample_fmt));
memcpy(
samples + (audio_input_position
* (av_get_bytes_per_sample(output_sample_fmt) /
av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) )
),
all_resampled_samples + samples_position,
diff * av_get_bytes_per_sample(output_sample_fmt)
);
// Increment counters
audio_input_position += diff;
@@ -1631,7 +1652,14 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
AVFrame *frame_final = AV_ALLOCATE_FRAME();
AV_RESET_FRAME(frame_final);
if (av_sample_fmt_is_planar(audio_codec->sample_fmt)) {
ZmqLogger::Instance()->AppendDebugMethod("FFmpegWriter::write_audio_packets (2nd resampling for Planar formats)", "in_sample_fmt", output_sample_fmt, "out_sample_fmt", audio_codec->sample_fmt, "in_sample_rate", info.sample_rate, "out_sample_rate", info.sample_rate, "in_channels", info.channels, "out_channels", info.channels);
ZmqLogger::Instance()->AppendDebugMethod("FFmpegWriter::write_audio_packets (2nd resampling for Planar formats)",
"in_sample_fmt", output_sample_fmt,
"out_sample_fmt", audio_codec->sample_fmt,
"in_sample_rate", info.sample_rate,
"out_sample_rate", info.sample_rate,
"in_channels", info.channels,
"out_channels", info.channels
);
// setup resample context
if (!avr_planar) {
@@ -1654,31 +1682,38 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
// Create a new array
final_samples_planar = (int16_t *) av_malloc(
sizeof(int16_t) * audio_frame->nb_samples * info.channels * (av_get_bytes_per_sample(output_sample_fmt) / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)));
sizeof(int16_t) * audio_frame->nb_samples * info.channels
* (av_get_bytes_per_sample(output_sample_fmt) /
av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) )
);
// Copy audio into buffer for frame
memcpy(final_samples_planar, samples, audio_frame->nb_samples * info.channels * av_get_bytes_per_sample(output_sample_fmt));
// Fill input frame with sample data
avcodec_fill_audio_frame(audio_frame, info.channels, output_sample_fmt, (uint8_t *) final_samples_planar,
audio_encoder_buffer_size, 0);
avcodec_fill_audio_frame(audio_frame, info.channels, output_sample_fmt,
(uint8_t *) final_samples_planar, audio_encoder_buffer_size, 0);
// Create output frame (and allocate arrays)
frame_final->nb_samples = audio_input_frame_size;
av_samples_alloc(frame_final->data, frame_final->linesize, info.channels, frame_final->nb_samples, audio_codec->sample_fmt, 0);
av_samples_alloc(frame_final->data, frame_final->linesize, info.channels,
frame_final->nb_samples, audio_codec->sample_fmt, 0);
// Convert audio samples
int nb_samples = SWR_CONVERT(avr_planar, // audio resample context
frame_final->data, // output data pointers
frame_final->linesize[0], // output plane size, in bytes. (0 if unknown)
frame_final->nb_samples, // maximum number of samples that the output buffer can hold
audio_frame->data, // input data pointers
audio_frame->linesize[0], // input plane size, in bytes (0 if unknown)
audio_frame->nb_samples); // number of input samples to convert
int nb_samples = SWR_CONVERT(
avr_planar, // audio resample context
frame_final->data, // output data pointers
frame_final->linesize[0], // output plane size, in bytes. (0 if unknown)
frame_final->nb_samples, // maximum number of samples that the output buffer can hold
audio_frame->data, // input data pointers
audio_frame->linesize[0], // input plane size, in bytes (0 if unknown)
audio_frame->nb_samples // number of input samples to convert
);
// Copy audio samples over original samples
if (nb_samples > 0)
memcpy(samples, frame_final->data[0], nb_samples * av_get_bytes_per_sample(audio_codec->sample_fmt) * info.channels);
memcpy(samples, frame_final->data[0],
nb_samples * av_get_bytes_per_sample(audio_codec->sample_fmt) * info.channels);
// deallocate AVFrame
av_freep(&(audio_frame->data[0]));
@@ -1690,7 +1725,10 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
} else {
// Create a new array
final_samples = (int16_t *) av_malloc(
sizeof(int16_t) * audio_input_position * (av_get_bytes_per_sample(audio_codec->sample_fmt) / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)));
sizeof(int16_t) * audio_input_position
* (av_get_bytes_per_sample(audio_codec->sample_fmt) /
av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) )
);
// Copy audio into buffer for frame
memcpy(final_samples, samples, audio_input_position * av_get_bytes_per_sample(audio_codec->sample_fmt));
@@ -1699,8 +1737,8 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
frame_final->nb_samples = audio_input_frame_size;
// Fill the final_frame AVFrame with audio (non planar)
avcodec_fill_audio_frame(frame_final, audio_codec->channels, audio_codec->sample_fmt, (uint8_t *) final_samples,
audio_encoder_buffer_size, 0);
avcodec_fill_audio_frame(frame_final, audio_codec->channels, audio_codec->sample_fmt,
(uint8_t *) final_samples, audio_encoder_buffer_size, 0);
}
// Increment PTS (in samples)
@@ -1872,7 +1910,10 @@ void FFmpegWriter::process_video_packet(std::shared_ptr<Frame> frame) {
} else
#endif // HAVE_HW_ACCEL
{
frame_final = allocate_avframe((AVPixelFormat)(video_st->codecpar->format), info.width, info.height, &bytes_final, NULL);
frame_final = allocate_avframe(
(AVPixelFormat)(video_st->codecpar->format),
info.width, info.height, &bytes_final, NULL
);
}
#else
AVFrame *frame_final = allocate_avframe(video_codec->pix_fmt, info.width, info.height, &bytes_final, NULL);
@@ -1884,7 +1925,7 @@ void FFmpegWriter::process_video_packet(std::shared_ptr<Frame> frame) {
// Resize & convert pixel format
sws_scale(scaler, frame_source->data, frame_source->linesize, 0,
source_image_height, frame_final->data, frame_final->linesize);
source_image_height, frame_final->data, frame_final->linesize);
// Add resized AVFrame to av_frames map
#pragma omp critical (av_frames_section)