C++ shared_ptr::SampleRate方法代碼示例

// Resample audio and map channels (if needed)
void FrameMapper::ResampleMappedAudio(tr1::shared_ptr<Frame> frame, long int original_frame_number)
{
	// Init audio buffers / variables
	int total_frame_samples = 0;
	int channels_in_frame = frame->GetAudioChannelsCount();
	int sample_rate_in_frame = frame->SampleRate();
	int samples_in_frame = frame->GetAudioSamplesCount();
	ChannelLayout channel_layout_in_frame = frame->ChannelsLayout();

	AppendDebugMethod("FrameMapper::ResampleMappedAudio", "frame->number", frame->number, "original_frame_number", original_frame_number, "channels_in_frame", channels_in_frame, "samples_in_frame", samples_in_frame, "sample_rate_in_frame", sample_rate_in_frame, "", -1);

	// 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));


	// Deallocate float array
	delete[] frame_samples_float;
	frame_samples_float = NULL;

	AppendDebugMethod("FrameMapper::ResampleMappedAudio (got sample data from frame)", "frame->number", frame->number, "total_frame_samples", total_frame_samples, "target channels", info.channels, "channels_in_frame", channels_in_frame, "target sample_rate", info.sample_rate, "samples_in_frame", samples_in_frame);


	// Create input frame (and allocate arrays)
	AVFrame *audio_frame = AV_ALLOCATE_FRAME();
	AV_RESET_FRAME(audio_frame);
	audio_frame->nb_samples = total_frame_samples / channels_in_frame;

	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);

	if (error_code < 0)
	{
		AppendDebugMethod("FrameMapper::ResampleMappedAudio ERROR [" + (string)av_err2str(error_code) + "]", "error_code", error_code, "", -1, "", -1, "", -1, "", -1, "", -1);
		throw ErrorEncodingVideo("Error while resampling audio in frame mapper", frame->number);
	}

	// Update total samples & input frame size (due to bigger or smaller data types)
	total_frame_samples = Frame::GetSamplesPerFrame(frame->number, target, info.sample_rate, info.channels);

	AppendDebugMethod("FrameMapper::ResampleMappedAudio (adjust # of samples)", "total_frame_samples", total_frame_samples, "info.sample_rate", info.sample_rate, "sample_rate_in_frame", sample_rate_in_frame, "info.channels", info.channels, "channels_in_frame", channels_in_frame, "original_frame_number", original_frame_number);

	// Create output frame (and allocate arrays)
	AVFrame *audio_converted = AV_ALLOCATE_FRAME();
	AV_RESET_FRAME(audio_converted);
	audio_converted->nb_samples = total_frame_samples;
	av_samples_alloc(audio_converted->data, audio_converted->linesize, info.channels, total_frame_samples, AV_SAMPLE_FMT_S16, 0);

	AppendDebugMethod("FrameMapper::ResampleMappedAudio (preparing for resample)", "in_sample_fmt", AV_SAMPLE_FMT_S16, "out_sample_fmt", AV_SAMPLE_FMT_S16, "in_sample_rate", sample_rate_in_frame, "out_sample_rate", info.sample_rate, "in_channels", channels_in_frame, "out_channels", info.channels);

	int nb_samples = 0;
	// 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
//.........這裏部分代碼省略.........