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conform audio to planar in the decoder

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This commit is contained in:
itsmattkc
2021-10-13 16:01:26 -07:00
parent 2eba9ebb4b
commit 4173eb3c49
14 changed files with 320 additions and 109 deletions
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app/codec/ffmpeg/ffmpegdecoder.cpp
+26 -15
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@@ -38,6 +38,7 @@ extern "C" {
#include <QThread>
#include <QtConcurrent/QtConcurrent>
#include "codec/planarfiledevice.h"
#include "common/define.h"
#include "common/ffmpegutils.h"
#include "common/filefunctions.h"
@@ -437,7 +438,7 @@ QString FFmpegDecoder::FFmpegError(int error_code)
return QStringLiteral("%1 %2").arg(QString::number(error_code), err);
}
bool FFmpegDecoder::ConformAudioInternal(const QString &filename, const AudioParams &params, const QAtomicInt *cancelled)
bool FFmpegDecoder::ConformAudioInternal(const QVector<QString> &filenames, const AudioParams &params, const QAtomicInt *cancelled)
{
// Iterate through each audio frame and extract the PCM data
@@ -454,7 +455,7 @@ bool FFmpegDecoder::ConformAudioInternal(const QString &filename, const AudioPar
// Create resampling context
SwrContext* resampler = swr_alloc_set_opts(nullptr,
params.channel_layout(),
FFmpegUtils::GetFFmpegSampleFormat(params.format()),
FFmpegUtils::GetFFmpegSampleFormat(params.format(), true),
params.sample_rate(),
channel_layout,
static_cast<AVSampleFormat>(instance_.avstream()->codecpar->format),
@@ -464,8 +465,6 @@ bool FFmpegDecoder::ConformAudioInternal(const QString &filename, const AudioPar
swr_init(resampler);
QFile wave_out(filename);
AVPacket* pkt = av_packet_alloc();
AVFrame* frame = av_frame_alloc();
int ret;
@@ -481,7 +480,12 @@ bool FFmpegDecoder::ConformAudioInternal(const QString &filename, const AudioPar
}
}
if (wave_out.open(QFile::WriteOnly)) {
PlanarFileDevice wave_out;
if (wave_out.open(filenames, QFile::WriteOnly)) {
int nb_channels = params.channel_count();
SampleBuffer data;
data.set_audio_params(params);
while (true) {
// Check if we have a `cancelled` ptr and its value
if (cancelled && *cancelled) {
@@ -505,15 +509,30 @@ bool FFmpegDecoder::ConformAudioInternal(const QString &filename, const AudioPar
// Allocate buffers
int nb_samples = swr_get_out_samples(resampler, frame->nb_samples);
char* data = new char[params.samples_to_bytes(nb_samples)];
int nb_bytes_per_channel = params.samples_to_bytes(nb_samples) / nb_channels;
data.set_sample_count(nb_bytes_per_channel);
data.allocate();
// Resample audio to our destination parameters
nb_samples = swr_convert(resampler,
reinterpret_cast<uint8_t**>(&data),
reinterpret_cast<uint8_t**>(data.to_raw_ptrs()),
nb_samples,
const_cast<const uint8_t**>(frame->data),
frame->nb_samples);
// If no error, write to files
if (nb_samples > 0) {
// Update byte count for the number of samples we actually received
nb_bytes_per_channel = params.samples_to_bytes(nb_samples) / nb_channels;
// Write to files
wave_out.write(const_cast<const char**>(reinterpret_cast<char**>(data.to_raw_ptrs())), nb_bytes_per_channel);
}
// Free buffer
data.destroy();
// Handle error now after freeing
if (nb_samples < 0) {
char err_str[512];
av_strerror(nb_samples, err_str, 512);
@@ -521,14 +540,6 @@ bool FFmpegDecoder::ConformAudioInternal(const QString &filename, const AudioPar
break;
}
// Write packed WAV data to the disk cache
wave_out.write(data, params.samples_to_bytes(nb_samples));
// If we allocated an output for the resampler, delete it here
if (data != reinterpret_cast<char*>(frame->data[0])) {
delete [] data;
}
SignalProcessingProgress(frame->pts, duration);
}