Method FillBufferDifferentRate

src/audio_resampler.cpp:498–619

}

int AudioResampler::FillBufferDifferentRate(uint8_t* buffer, int length) {
	const int input_samplesize = AudioDecoder::GetSamplesizeForFormat(input_format);
	const int output_samplesize = AudioDecoder::GetSamplesizeForFormat(output_format);
	//The buffer size has to be a multiple of a frame
	const int buffer_size=sizeof(internal_buffer) - sizeof(internal_buffer)%(nr_of_channels*((input_samplesize>output_samplesize) ? input_samplesize : output_samplesize));

	int total_output_frames = length / (output_samplesize*nr_of_channels);
	int amount_of_samples_to_read = 0;
	int amount_of_samples_read = 0;

	uint8_t * advanced_input_buffer = internal_buffer;
	int unused_frames = 0;
	int empty_buffer_space = 0;
	int error = 0;

	#ifdef HAVE_LIBSPEEXDSP
		spx_uint32_t numerator = 0;
		spx_uint32_t denominator = 0;
	#endif

	while (total_output_frames > 0) {
		//Calculate how much frames of the last cycle are unused - to reuse them
		unused_frames = conversion_data.input_frames - conversion_data.input_frames_used;
		empty_buffer_space = buffer_size / output_samplesize - unused_frames*nr_of_channels;

		advanced_input_buffer = internal_buffer;

		//If there is still unused data in the input_buffer order it to the front
		for (int i = 0; i < unused_frames*nr_of_channels*output_samplesize; i++) {
			*advanced_input_buffer = *(advanced_input_buffer + empty_buffer_space*output_samplesize);
			advanced_input_buffer++;
		}
		//advanced_input_buffer is now offset to the first frame of new data!

		//ensure that the input buffer is not able to overrun
		amount_of_samples_to_read = (input_samplesize > output_samplesize) ? (empty_buffer_space*output_samplesize) / input_samplesize : empty_buffer_space;

		//Read as many frames as needed to refill the buffer (filled after the conversion to float)
		if (amount_of_samples_to_read != 0) {
			switch (output_format) {
				case AudioDecoder::Format::F32: amount_of_samples_read = DecodeAndConvertFloat(wrapped_decoder.get(), advanced_input_buffer, amount_of_samples_to_read, input_samplesize, input_format); break;
			#ifdef HAVE_LIBSPEEXDSP
				case AudioDecoder::Format::S16:  amount_of_samples_read = DecodeAndConvertInt16(wrapped_decoder.get(), advanced_input_buffer, amount_of_samples_to_read, input_samplesize, input_format); break;
			#endif
				default: error_message = "internal error: output_format is not convertable"; return ERROR;
			}
			if (amount_of_samples_read < 0) {
				error_message = wrapped_decoder->GetError();
				return amount_of_samples_read; //error occured
			}
		}
		//Now we have a prepared full buffer of converted values

		//Prepare the source data
		conversion_data.input_frames = amount_of_samples_read / nr_of_channels + unused_frames;
		conversion_data.output_frames = total_output_frames;

		#if defined(HAVE_LIBSPEEXDSP)