C++ shared_ptr::ResizeAudio方法代码示例

//.........这里部分代码省略.........
	// Force the audio resampling to happen in order (1st thread to last thread), so the waveform
	// is smooth and continuous.
	#pragma omp ordered
	{
		// setup resample context
		if (!avr) {
			avr = avresample_alloc_context();
			av_opt_set_int(avr,  "in_channel_layout", channel_layout_in_frame, 0);
			av_opt_set_int(avr, "out_channel_layout", info.channel_layout, 0);
			av_opt_set_int(avr,  "in_sample_fmt",     AV_SAMPLE_FMT_S16,     0);
			av_opt_set_int(avr, "out_sample_fmt",     AV_SAMPLE_FMT_S16,     0);
			av_opt_set_int(avr,  "in_sample_rate",    sample_rate_in_frame,    0);
			av_opt_set_int(avr, "out_sample_rate",    info.sample_rate,    0);
			av_opt_set_int(avr,  "in_channels",       channels_in_frame,    0);
			av_opt_set_int(avr, "out_channels",       info.channels,    0);
			avresample_open(avr);
		}

		// Convert audio samples
		nb_samples = avresample_convert(avr, 	// audio resample context
				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
				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(resampled_samples, audio_converted->data[0], (nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * info.channels));

	// Free frames
	free(audio_frame->data[0]); // TODO: Determine why av_free crashes on Windows
	AV_FREE_FRAME(&audio_frame);
	av_free(audio_converted->data[0]);
	AV_FREE_FRAME(&audio_converted);
	frame_samples = NULL;

	// Resize the frame to hold the right # of channels and samples
	int channel_buffer_size = nb_samples;
	frame->ResizeAudio(info.channels, channel_buffer_size, info.sample_rate, info.channel_layout);

	AppendDebugMethod("FrameMapper::ResampleMappedAudio (Audio successfully resampled)", "nb_samples", nb_samples, "total_frame_samples", total_frame_samples, "info.sample_rate", info.sample_rate, "channels_in_frame", channels_in_frame, "info.channels", info.channels, "info.channel_layout", info.channel_layout);

	// Array of floats (to hold samples for each channel)
	float *channel_buffer = new float[channel_buffer_size];

	// Divide audio into channels. Loop through each channel
	for (int channel_filter = 0; channel_filter < info.channels; channel_filter++)
	{
		// Init array
		for (int z = 0; z < channel_buffer_size; z++)
			channel_buffer[z] = 0.0f;

		// Loop through all samples and add them to our Frame based on channel.
		// Toggle through each channel number, since channel data is stored like (left right left right)
		int channel = 0;
		int position = 0;
		for (int sample = 0; sample < (nb_samples * info.channels); sample++)
		{
			// 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] = resampled_samples[sample] * (1.0f / (1 << 15));

				// Increment audio position
				position++;
			}

			// increment channel (if needed)
			if ((channel + 1) < info.channels)
				// move to next channel
				channel ++;
			else
				// reset channel
				channel = 0;
		}

		// Add samples to frame for this channel
		frame->AddAudio(true, channel_filter, 0, channel_buffer, position, 1.0f);

		AppendDebugMethod("FrameMapper::ResampleMappedAudio (Add audio to channel)", "number of samples", position, "channel_filter", channel_filter, "", -1, "", -1, "", -1, "", -1);
	}

	// Update frame's audio meta data
	frame->SampleRate(info.sample_rate);
	frame->ChannelsLayout(info.channel_layout);

	// clear channel buffer
	delete[] channel_buffer;
	channel_buffer = NULL;

	// Delete arrays
	delete[] resampled_samples;
	resampled_samples = NULL;
}

// Process a new layer of video or audio
void Timeline::add_layer(tr1::shared_ptr<Frame> new_frame, Clip* source_clip, long int clip_frame_number, long int timeline_frame_number, bool is_top_clip)
{
	// Get the clip's frame & image
	tr1::shared_ptr<Frame> source_frame = GetOrCreateFrame(source_clip, clip_frame_number);

	// No frame found... so bail
	if (!source_frame)
		return;

	// Debug output
	AppendDebugMethod("Timeline::add_layer", "new_frame->number", new_frame->number, "clip_frame_number", clip_frame_number, "timeline_frame_number", timeline_frame_number, "", -1, "", -1, "", -1);

	/* REPLACE IMAGE WITH WAVEFORM IMAGE (IF NEEDED) */
	if (source_clip->Waveform())
	{
		// Debug output
		AppendDebugMethod("Timeline::add_layer (Generate Waveform Image)", "source_frame->number", source_frame->number, "source_clip->Waveform()", source_clip->Waveform(), "clip_frame_number", clip_frame_number, "", -1, "", -1, "", -1);

		// Get the color of the waveform
		int red = source_clip->wave_color.red.GetInt(clip_frame_number);
		int green = source_clip->wave_color.green.GetInt(clip_frame_number);
		int blue = source_clip->wave_color.blue.GetInt(clip_frame_number);
		int alpha = source_clip->wave_color.alpha.GetInt(clip_frame_number);

		// Generate Waveform Dynamically (the size of the timeline)
		tr1::shared_ptr<QImage> source_image = source_frame->GetWaveform(info.width, info.height, red, green, blue, alpha);
		source_frame->AddImage(tr1::shared_ptr<QImage>(source_image));
	}

	/* Apply effects to the source frame (if any). If multiple clips are overlapping, only process the
	 * effects on the top clip. */
	if (is_top_clip)
		source_frame = apply_effects(source_frame, timeline_frame_number, source_clip->Layer());

	// Declare an image to hold the source frame's image
	tr1::shared_ptr<QImage> source_image;

	/* COPY AUDIO - with correct volume */
	if (source_clip->Reader()->info.has_audio) {

		// Debug output
		AppendDebugMethod("Timeline::add_layer (Copy Audio)", "source_clip->Reader()->info.has_audio", source_clip->Reader()->info.has_audio, "source_frame->GetAudioChannelsCount()", source_frame->GetAudioChannelsCount(), "info.channels", info.channels, "clip_frame_number", clip_frame_number, "timeline_frame_number", timeline_frame_number, "", -1);

		if (source_frame->GetAudioChannelsCount() == info.channels)
			for (int channel = 0; channel < source_frame->GetAudioChannelsCount(); channel++)
			{
				float initial_volume = 1.0f;
				float previous_volume = source_clip->volume.GetValue(clip_frame_number - 1); // previous frame's percentage of volume (0 to 1)
				float volume = source_clip->volume.GetValue(clip_frame_number); // percentage of volume (0 to 1)

				// If no ramp needed, set initial volume = clip's volume
				if (isEqual(previous_volume, volume))
					initial_volume = volume;

				// Apply ramp to source frame (if needed)
				if (!isEqual(previous_volume, volume))
					source_frame->ApplyGainRamp(channel, 0, source_frame->GetAudioSamplesCount(), previous_volume, volume);

				// TODO: Improve FrameMapper (or Timeline) to always get the correct number of samples per frame.
				// Currently, the ResampleContext sometimes leaves behind a few samples for the next call, and the
				// number of samples returned is variable... and does not match the number expected.
				// This is a crude solution at best. =)
				if (new_frame->GetAudioSamplesCount() != source_frame->GetAudioSamplesCount())
					// Force timeline frame to match the source frame
					new_frame->ResizeAudio(info.channels, source_frame->GetAudioSamplesCount(), info.sample_rate, info.channel_layout);

				// Copy audio samples (and set initial volume).  Mix samples with existing audio samples.  The gains are added together, to
				// be sure to set the gain's correctly, so the sum does not exceed 1.0 (of audio distortion will happen).
				new_frame->AddAudio(false, channel, 0, source_frame->GetAudioSamples(channel), source_frame->GetAudioSamplesCount(), initial_volume);

			}
		else
			// Debug output
			AppendDebugMethod("Timeline::add_layer (No Audio Copied - Wrong # of Channels)", "source_clip->Reader()->info.has_audio", source_clip->Reader()->info.has_audio, "source_frame->GetAudioChannelsCount()", source_frame->GetAudioChannelsCount(), "info.channels", info.channels, "clip_frame_number", clip_frame_number, "timeline_frame_number", timeline_frame_number, "", -1);

	}

	// Skip out if only an audio frame
	if (!source_clip->Waveform() && !source_clip->Reader()->info.has_video)
		// Skip the rest of the image processing for performance reasons
		return;

	// Debug output
	AppendDebugMethod("Timeline::add_layer (Get Source Image)", "source_frame->number", source_frame->number, "source_clip->Waveform()", source_clip->Waveform(), "clip_frame_number", clip_frame_number, "", -1, "", -1, "", -1);

	// Get actual frame image data
	source_image = source_frame->GetImage();

	// Get some basic image properties
	int source_width = source_image->width();
	int source_height = source_image->height();

	/* ALPHA & OPACITY */
	if (source_clip->alpha.GetValue(clip_frame_number) != 1.0)
	{
		float alpha = source_clip->alpha.GetValue(clip_frame_number);

		// Get source image's pixels
		unsigned char *pixels = (unsigned char *) source_image->bits();
//.........这里部分代码省略.........