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Added in JUCE audio resampling, and have lots of resampling debug code still in place.

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This commit is contained in:
Jonathan Thomas
2012-07-30 02:37:19 -05:00
parent 8387f503a6
commit efb38cc9d8
5 changed files with 165 additions and 24 deletions
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69
src/FFmpegWriter.cpp
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@@ -314,8 +314,8 @@ AVStream* FFmpegWriter::add_audio_stream()
// Check for valid timebase
if (c->time_base.den == 0 || c->time_base.num == 0)
{
c->time_base.num = 1;
c->time_base.den = 90000;
c->time_base.num = st->time_base.num;
c->time_base.den = st->time_base.den;
}
// Set valid sample rate (or throw error)
@@ -334,6 +334,7 @@ AVStream* FFmpegWriter::add_audio_stream()
// Set sample rate
c->sample_rate = info.sample_rate;
// Set a valid number of channels (or throw error)
int channel_layout = info.channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
if (codec->channel_layouts) {
@@ -351,6 +352,7 @@ AVStream* FFmpegWriter::add_audio_stream()
// Set valid channel layout
c->channel_layout = channel_layout;
// Choose a valid sample_fmt
if (codec->sample_fmts) {
for (int i = 0; codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; i++)
@@ -504,22 +506,31 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
c = audio_st->codec;
// Get the audio details from this frame
int sample_rate_in_frame = frame->GetAudioSamplesRate();
int samples_in_frame = frame->GetAudioSamplesCount();
int sample_rate_in_frame = info.sample_rate; // resampling happens when getting the interleaved audio samples below
int samples_in_frame = frame->GetAudioSamplesCount(); // this is updated if resampling happens
int channels_in_frame = frame->GetAudioChannelsCount();
int total_frame_samples = samples_in_frame * channels_in_frame;
int remaining_frame_samples = total_frame_samples;
// Get audio sample array
float* frame_samples_float = frame->GetAudioSamples();
float* frame_samples_float = frame->GetInterleavedAudioSamples(info.sample_rate, &samples_in_frame);
int16_t* frame_samples = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
int samples_position = 0;
// Calculate total samples
int total_frame_samples = samples_in_frame * channels_in_frame;
int remaining_frame_samples = 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));
// // DEBUG
// if (frame->number == 1)
// {
// cout << frame_samples[s] << endl;
// }
}
// Re-sample audio samples (if needed)
@@ -530,7 +541,7 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
ReSampleContext *resampleCtx = av_audio_resample_init(
info.channels, channels_in_frame,
info.sample_rate, sample_rate_in_frame,
c->sample_fmt, AV_SAMPLE_FMT_S16, 0, 0, 0, 0.0f);
c->sample_fmt, AV_SAMPLE_FMT_S16, 16, 10, 0, 0.0f);
if (!resampleCtx)
throw InvalidCodec("Failed to convert audio samples for encoding.", path);
@@ -539,6 +550,14 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
total_frame_samples = audio_resample(resampleCtx, (short *) converted_audio, (short *) frame_samples, total_frame_samples);
remaining_frame_samples = total_frame_samples;
// DEBUG
// cout << endl << endl << endl << "END OF RESAMPLE DATA:" << endl;
// if (frame->number == 1)
// {
// for (int s = total_frame_samples - 10; s < (total_frame_samples + 200); s++)
// cout << converted_audio[s] << endl;
// }
// Copy audio samples over original samples
memcpy(frame_samples, converted_audio, total_frame_samples * av_get_bytes_per_sample(c->sample_fmt));
@@ -547,6 +566,14 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
}
}
// // DEBUG
// cout << endl << endl << endl << "AFTER RESCALE" << endl;
// if (frame->number == 1)
// {
// for (int s = 0; s < total_frame_samples; s++)
// cout << frame_samples[s] << endl;
// }
// Loop until no more samples
while (remaining_frame_samples > 0) {
// Get remaining samples needed for this packet
@@ -567,7 +594,7 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
// Copy samples into input buffer (and convert to 16 bit int)
for (int s = 0; s <= diff; s++)
for (int s = 0; s < diff; s++)
{
// Translate sample value and copy into buffer
samples[audio_input_position] = frame_samples[samples_position];
@@ -577,20 +604,32 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
samples_position++;
remaining_frame_samples--;
remaining_packet_samples--;
// DEBUG
// if (frame->number == 1 && s > 2800)
// {
// cout << frame_samples[s] << endl;
// }
// if (frame->number == 2 && s < 100)
// {
// cout << frame_samples[s] << endl;
// }
}
// if (frame->number == 2 || frame->number == 3)
// {
// for (int z = 0; z < diff; z++)
// cout << frame_samples[z] << endl;
// }
// Do we have enough samples to proceed?
if (audio_input_position < audio_input_frame_size)
// Not enough samples to encode... so wait until the next frame
break;
cout << "NOW ENCODE FILE!" << endl;
// if (frame->number == 5)
// {
// for (int z = 0; z < audio_input_frame_size; z++)
// cout << samples[z] << endl;
// }
// Set packet properties
pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, (short *) samples);
@@ -601,6 +640,8 @@ void FFmpegWriter::write_audio_packet(Frame* frame)
pkt.stream_index = audio_st->index;
pkt.data = audio_outbuf;
cout << "Final encode: frame " << frame->number << ", pkt.pts: " << pkt.pts << endl;
/* write the compressed frame in the media file */
if (av_interleaved_write_frame(oc, &pkt) != 0)
throw ErrorEncodingAudio("Error while writing audio frame", frame->number);