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Fixed a bug where audio was not being resampled to the correct sample format. Also fixed a bug where the audio buffer was not being incremented correctly, and it was duplicating audio samples. There are still a few audio bugs, but it's mostly working now.

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This commit is contained in:
Jonathan Thomas
2011-12-18 02:29:34 -06:00
parent b4c7a6ce81
commit da289f93bf
3 changed files with 53 additions and 22 deletions
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70
src/FFmpegReader.cpp
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@@ -5,7 +5,8 @@ using namespace openshot;
FFmpegReader::FFmpegReader(string path) throw(InvalidFile, NoStreamsFound, InvalidCodec)
: last_video_frame(0), last_audio_frame(0), is_seeking(0), seeking_pts(0), seeking_frame(0),
audio_pts_offset(99999), video_pts_offset(99999), working_cache(50), path(path),
is_video_seek(true), check_interlace(false), check_fps(false), init_settings(false), enable_seek(true) {
is_video_seek(true), check_interlace(false), check_fps(false), init_settings(false),
enable_seek(true), resampleCtx(NULL) {
// Open the file (if possible)
Open();
@@ -116,6 +117,8 @@ void FFmpegReader::Close()
avcodec_close(pCodecCtx);
if (info.has_audio)
avcodec_close(aCodecCtx);
if (resampleCtx)
audio_resample_close(resampleCtx);
// Close the video file
av_close_input_file(pFormatCtx);
@@ -480,13 +483,45 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
}
// Calculate total number of samples
packet_samples += (buf_size / sizeof(int16_t));
packet_samples += (buf_size / (av_get_bits_per_sample_format(aCodecCtx->sample_fmt) / 8));
// process samples...
packet.data += used;
packet.size -= used;
}
// Re-sample audio samples (if needed)
if(aCodecCtx->sample_fmt != SAMPLE_FMT_S16) {
// Audio needs to be converted
if(!resampleCtx)
// Create an audio resample context object (used to convert audio samples)
resampleCtx = av_audio_resample_init(
info.channels,
info.channels,
info.sample_rate,
info.sample_rate,
SAMPLE_FMT_S16,
aCodecCtx->sample_fmt,
0, 0, 0, 0.0f);
if (!resampleCtx)
throw InvalidCodec("Failed to convert audio samples to 16 bit (SAMPLE_FMT_S16).", path);
else
{
// create a new array (to hold the re-sampled audio)
int16_t converted_audio[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
// Re-sample audio
audio_resample(resampleCtx, (short *)&converted_audio, (short *)&audio_buf, packet_samples);
// Copy audio samples over original samples
memcpy(&audio_buf, (uint8_t *)&converted_audio, packet_samples);
}
}
for (int channel_filter = 0; channel_filter < info.channels; channel_filter++)
{
// Array of floats (to hold samples for each channel)
@@ -508,24 +543,9 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
// Only add samples for current channel
if (channel_filter == channel)
{
// Add sample (convert from (-32768 to 32768) to (-1.0 to 1.0))
channel_buffer[position] = audio_buf[sample] * (1.0f / (1 << 15));
// Experimental audio conversion
// switch (aCodecCtx->sample_fmt)
// {
// case SAMPLE_FMT_U8:
// channel_buffer[position] = audio_buf[sample] * (1.0f / (1 << 7));
// case SAMPLE_FMT_S16:
// channel_buffer[position] = audio_buf[sample] * (1.0f / (1 << 15));
// case SAMPLE_FMT_S32:
// channel_buffer[position] = audio_buf[sample] * (1.0f / (1 << 31));
// case SAMPLE_FMT_FLT:
// channel_buffer[position] = audio_buf[sample];
// case SAMPLE_FMT_DBL:
// channel_buffer[position] = audio_buf[sample];
// }
//cout << channel << "\t" << channel_buffer[position] << endl;
// Increment audio position
position++;
@@ -546,6 +566,7 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
// Loop through samples, and add them to the correct frames
int start = starting_sample;
int remaining_samples = channel_buffer_size;
float *iterate_channel_buffer = channel_buffer;
while (remaining_samples > 0)
{
// Create or get frame object
@@ -558,14 +579,22 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
samples = remaining_samples;
// Add samples for current channel to the frame
f.AddAudio(channel_filter, start, channel_buffer, samples, 1.0f);
cout << "starting_frame_number: " << starting_frame_number << endl;
cout << "channel_filter: " << channel_filter << endl;
cout << "samples: " << samples << endl;
cout << "remaining_samples: " << remaining_samples << endl;
f.AddAudio(channel_filter, start, iterate_channel_buffer, samples, 1.0f);
// Update working cache
working_cache.Add(starting_frame_number, f);
working_cache.Add(f.number, f);
// Decrement remaining samples
remaining_samples -= samples;
// Increment buffer (to next set of samples)
if (remaining_samples > 0)
iterate_channel_buffer += samples;
// Increment frame number
starting_frame_number++;
@@ -576,6 +605,7 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
// clear channel buffer
delete channel_buffer;
channel_buffer = NULL;
iterate_channel_buffer = NULL;
}
}