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Merge pull request #466 from SuslikV/float-to-int16

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Fix distortion during mix of audio
This commit is contained in:
Frank Dana
2020-03-26 19:51:32 -04:00
committed by GitHub
parent fda135725b 455b6e92bf
commit c38c55fbfa
4 changed files with 37 additions and 13 deletions
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5
src/FFmpegReader.cpp
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@@ -1627,9 +1627,8 @@ void FFmpegReader::ProcessAudioPacket(int64_t requested_frame, int64_t target_fr
else
partial_frame = true;
// Add samples for current channel to the frame. Reduce the volume to 98%, to prevent
// some louder samples from maxing out at 1.0 (not sure why this happens)
f->AddAudio(true, channel_filter, start, iterate_channel_buffer, samples, 0.98f);
// Add samples for current channel to the frame.
f->AddAudio(true, channel_filter, start, iterate_channel_buffer, samples, 1.0f);
// Debug output
ZmqLogger::Instance()->AppendDebugMethod("FFmpegReader::ProcessAudioPacket (f->AddAudio)", "frame", starting_frame_number, "start", start, "samples", samples, "channel", channel_filter, "partial_frame", partial_frame, "samples_per_frame", samples_per_frame);

20
src/FFmpegWriter.cpp
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@@ -1525,11 +1525,23 @@ void FFmpegWriter::write_audio_packets(bool is_final) {
// Calculate total samples
total_frame_samples = samples_in_frame * channels_in_frame;
// 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
all_queued_samples[frame_position] = int(frame_samples_float[s] * (1 << 15));
// 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)
conv = max16;
else if (valF < min16)
conv = min16;
else
conv = int(valF + 32768.5) - 32768; // +0.5 is for rounding
// Copy into buffer
all_queued_samples[frame_position] = conv;
}
// Deallocate float array
delete[] frame_samples_float;

21
src/FrameMapper.cpp
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@@ -785,10 +785,23 @@ void FrameMapper::ResampleMappedAudio(std::shared_ptr<Frame> frame, int64_t orig
// Create a new array (to hold all S16 audio samples for the current queued frames)
int16_t* frame_samples = (int16_t*) av_malloc(sizeof(int16_t)*total_frame_samples);
// Translate audio sample values back to 16 bit integers
for (int s = 0; s < total_frame_samples; s++)
// Translate sample value and copy into buffer
frame_samples[s] = int(frame_samples_float[s] * (1 << 15));
// 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++) {
valF = frame_samples_float[s] * (1 << 15);
if (valF > max16)
conv = max16;
else if (valF < min16)
conv = min16;
else
conv = int(valF + 32768.5) - 32768; // +0.5 is for rounding
// Copy into buffer
frame_samples[s] = conv;
}
// Deallocate float array

4
tests/FFmpegReader_Tests.cpp
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@@ -76,8 +76,8 @@ TEST(FFmpegReader_Check_Audio_File)
CHECK_CLOSE(0.0f, samples[50], 0.00001);
CHECK_CLOSE(0.0f, samples[100], 0.00001);
CHECK_CLOSE(0.0f, samples[200], 0.00001);
CHECK_CLOSE(0.160781f, samples[230], 0.00001);
CHECK_CLOSE(-0.06125f, samples[300], 0.00001);
CHECK_CLOSE(0.16406f, samples[230], 0.00001);
CHECK_CLOSE(-0.06250f, samples[300], 0.00001);
// Close reader
r.Close();