diff --git a/src/FFmpegReader.cpp b/src/FFmpegReader.cpp
index 73e39b7c..5afcfbb2 100644
--- a/src/FFmpegReader.cpp
+++ b/src/FFmpegReader.cpp
@@ -995,7 +995,7 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
// Convert audio samples
nb_samples = avresample_convert(avr, // audio resample context
- audio_converted->data, // output data pointers
+ 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
diff --git a/src/FFmpegWriter.cpp b/src/FFmpegWriter.cpp
index 914fb585..5f3da953 100644
--- a/src/FFmpegWriter.cpp
+++ b/src/FFmpegWriter.cpp
@@ -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
}
diff --git a/src/FrameMapper.cpp b/src/FrameMapper.cpp
index 937a1495..58921678 100644
--- a/src/FrameMapper.cpp
+++ b/src/FrameMapper.cpp
@@ -548,6 +548,13 @@ void FrameMapper::ChangeMapping(Fraction target_fps, PulldownType target_pulldow
info.sample_rate = target_sample_rate;
info.channels = target_channels;
info.channel_layout = target_channel_layout;
+
+ // Deallocate resample buffer
+ if (avr) {
+ avresample_close(avr);
+ avresample_free(&avr);
+ avr = NULL;
+ }
}
// Resample audio and map channels (if needed)
@@ -563,23 +570,21 @@ void FrameMapper::ResampleMappedAudio(tr1::shared_ptr frame)
int samples_in_frame = frame->GetAudioSamplesCount();
ChannelLayout channel_layout_in_frame = frame->ChannelsLayout();
- // Create a new array (to hold all S16 audio samples, for the current queued frames
- int16_t* frame_samples = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
-
// 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;
+ // Create a new array (to hold all S16 audio samples for the current queued frames)
+ int16_t* frame_samples = new 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));
- //frame_samples[s] = s * 3;
// Deallocate float array
delete[] frame_samples_float;
@@ -593,8 +598,7 @@ void FrameMapper::ResampleMappedAudio(tr1::shared_ptr frame)
AVFrame *audio_frame = avcodec_alloc_frame();
avcodec_get_frame_defaults(audio_frame);
audio_frame->nb_samples = total_frame_samples / channels_in_frame;
- //audio_frame->sample_rate = frame->SampleRate();
- av_samples_alloc(audio_frame->data, audio_frame->linesize, channels_in_frame, audio_frame->nb_samples, AV_SAMPLE_FMT_S16, 0);
+ //av_samples_alloc(audio_frame->data, audio_frame->linesize, channels_in_frame, audio_frame->nb_samples, AV_SAMPLE_FMT_S16, 0);
int error_code = avcodec_fill_audio_frame(audio_frame, channels_in_frame, AV_SAMPLE_FMT_S16, (uint8_t *) frame_samples,
audio_frame->nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * channels_in_frame, 1);
@@ -614,7 +618,6 @@ void FrameMapper::ResampleMappedAudio(tr1::shared_ptr frame)
AVFrame *audio_converted = avcodec_alloc_frame();
avcodec_get_frame_defaults(audio_converted);
audio_converted->nb_samples = audio_frame->nb_samples;
- //audio_converted->sample_rate = info.sample_rate;
av_samples_alloc(audio_converted->data, audio_converted->linesize, info.channels, audio_frame->nb_samples, AV_SAMPLE_FMT_S16, 0);
#pragma omp critical (debug_output)
@@ -644,12 +647,16 @@ void FrameMapper::ResampleMappedAudio(tr1::shared_ptr frame)
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)
+ int16_t* resampled_samples = new int16_t[nb_samples * info.channels];
// Copy audio samples over original samples
- memcpy(frame_samples, audio_converted->data[0], nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * info.channels);
+ memcpy(resampled_samples, audio_converted->data[0], nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * info.channels);
// Free frames
+ av_freep(&audio_frame[0]);
avcodec_free_frame(&audio_frame); // TODO: Find a way to clear the memory inside this frame (memory leak)
+ frame_samples = NULL;
av_freep(&audio_converted[0]);
avcodec_free_frame(&audio_converted);
@@ -680,7 +687,7 @@ void FrameMapper::ResampleMappedAudio(tr1::shared_ptr frame)
if (channel_filter == channel)
{
// Add sample (convert from (-32768 to 32768) to (-1.0 to 1.0))
- channel_buffer[position] = frame_samples[sample] * (1.0f / (1 << 15));
+ channel_buffer[position] = resampled_samples[sample] * (1.0f / (1 << 15));
// Increment audio position
position++;
@@ -707,5 +714,6 @@ void FrameMapper::ResampleMappedAudio(tr1::shared_ptr frame)
channel_buffer = NULL;
// Delete arrays
- delete[] frame_samples;
+ delete[] resampled_samples;
+ resampled_samples = NULL;
}
diff --git a/src/examples/Example.cpp b/src/examples/Example.cpp
index 4e64ed72..084887b9 100644
--- a/src/examples/Example.cpp
+++ b/src/examples/Example.cpp
@@ -97,7 +97,7 @@ int main(int argc, char* argv[])
//r9.GetFrame(frame_number)->DisplayWaveform();
//f->DisplayWaveform();
//f->AddColor(r9.info.width, r9.info.height, "blue");
- //w9.WriteFrame(f);
+ w9.WriteFrame(f);
//frame++;
}
@@ -112,6 +112,7 @@ int main(int argc, char* argv[])
// Close timeline
r9.Close();
+ map.Close();
/* ---------------- */
cout << "happy ending" << endl;