fs2open.github.com/code/sound/ffmpeg/WaveFile.cpp

//
//

#include "WaveFile.h"
#include "FFmpegAudioReader.h"
#include "sound/ds.h"

// -- from parselo.cpp --
extern const char* stristr(const char* str, const char* substr);

namespace {
using namespace ffmpeg;

AudioProperties getAudioProps(AVStream* stream) {
	AudioProperties props;

#if LIBAVCODEC_VERSION_INT > AV_VERSION_INT(57, 24, 255)
	props.channels = stream->codecpar->channels;
	props.channel_layout = stream->codecpar->channel_layout;
	props.sample_rate = stream->codecpar->sample_rate;
	props.format = (AVSampleFormat)stream->codecpar->format;
#else
	props.channels = stream->codec->channels;
	props.channel_layout = stream->codec->channel_layout;
	props.sample_rate = stream->codec->sample_rate;
	props.format = stream->codec->sample_fmt;
#endif
	if (props.channel_layout == 0) {
		// Use a default channel layout value
		props.channel_layout = (uint64_t) av_get_default_channel_layout(props.channels);
	}

	return props;
}

AudioProperties getAdjustedAudioProps(const AudioProperties& baseProps) {
	AudioProperties adjusted;
	adjusted.sample_rate = baseProps.sample_rate; // Don't adjust sample rate

	adjusted.channel_layout = baseProps.channel_layout;
	if (av_get_channel_layout_nb_channels(baseProps.channel_layout) > 2) {
		adjusted.channel_layout = AV_CH_LAYOUT_STEREO;
	}
	adjusted.channels = av_get_channel_layout_nb_channels(adjusted.channel_layout);

	int max_bytes_per_sample;
	switch (Ds_sound_quality) {
		case DS_SQ_HIGH:
			max_bytes_per_sample = Ds_float_supported ? 4 : 2;
			break;

		case DS_SQ_MEDIUM:
			max_bytes_per_sample = 2;
			break;

		default:
			max_bytes_per_sample = 1;
			break;
	}

	// Determine the right format. Downsample if sound quality is low but don't increase size of samples if the source doesn't support it
	auto bytes_per_sample = std::min(av_get_bytes_per_sample(baseProps.format), max_bytes_per_sample);

	switch (bytes_per_sample) {
		case 1:
			adjusted.format = AV_SAMPLE_FMT_U8;
		case 2:
			adjusted.format = AV_SAMPLE_FMT_S16;
			break;
		case 4:
			adjusted.format = AV_SAMPLE_FMT_FLT;
			break;
		default:
			Assertion(false, "Unhandled switch value!");
			adjusted.format = AV_SAMPLE_FMT_NONE;
			break;
	}

	return adjusted;
}

SwrContext* getSWRContext(const AudioProperties& base, const AudioProperties& adjusted) {
	auto swr = swr_alloc();

	av_opt_set_int(swr, "in_channel_layout", base.channel_layout, 0);
	av_opt_set_int(swr, "in_sample_rate", base.sample_rate, 0);
	av_opt_set_int(swr, "in_sample_fmt", base.format, 0);

	av_opt_set_int(swr, "out_channel_layout", adjusted.channel_layout, 0);
	av_opt_set_int(swr, "out_sample_rate", adjusted.sample_rate, 0);
	av_opt_set_int(swr, "out_sample_fmt", adjusted.format, 0);

	swr_init(swr);

	return swr;
}
}

namespace ffmpeg
{
WaveFile::WaveFile() : m_resampleCtx(nullptr) {
	// Init data members
	m_audioProps = AudioProperties();
	m_ctx.reset();

	m_al_format = AL_FORMAT_MONO8;

	m_decodeFrame = av_frame_alloc();
}

WaveFile::~WaveFile() {
	av_frame_free(&m_decodeFrame);

	if (m_audioCodecCtx) {
		avcodec_close(m_audioCodecCtx);
#if LIBAVCODEC_VERSION_INT > AV_VERSION_INT(57, 24, 255)
		avcodec_free_context(&m_audioCodecCtx);
#endif
		m_audioCodecCtx = nullptr;
	}

	if (m_resampleCtx) {
		swr_free(&m_resampleCtx);
	}

	// Free memory
	m_ctx.reset();

}

// Open
bool WaveFile::Open(const char *pszFilename, bool keep_ext)
{
	using namespace libs::ffmpeg;

	size_t FileSize, FileOffset;
	char fullpath[MAX_PATH];
	char filename[MAX_FILENAME_LEN];

	// NOTE: we assume that the extension has already been stripped off if it was supposed to be!!
	strcpy_s( filename, pszFilename );

	try {
		int rc = -1;
		// if we are supposed to load the file as passed...
		if (keep_ext) {
			for (int i = 0; i < NUM_AUDIO_EXT; i++) {
				if (stristr(pszFilename, audio_ext_list[i])) {
					rc = i;
					break;
				}
			}

			// not a supported extension format ... somebody screwed up their tbls :)
			if (rc < 0) {
				throw FFmpegException("Unknown file extension.");
			}

			cf_find_file_location(pszFilename, CF_TYPE_ANY, sizeof(fullpath) - 1, fullpath, &FileSize, &FileOffset);
		}
		// ... otherwise we just find the best match
		else {
			rc = cf_find_file_location_ext(filename,
										   NUM_AUDIO_EXT,
										   audio_ext_list,
										   CF_TYPE_ANY,
										   sizeof(fullpath) - 1,
										   fullpath,
										   &FileSize,
										   &FileOffset);

			if (rc < 0) {
				throw FFmpegException("Unknown file extension.");
			}

			// set proper filename for later use (assumes that it doesn't already have an extension)
			strcat_s(filename, audio_ext_list[rc]);
		}

		auto cfp = cfopen_special(fullpath, "rb", FileSize, FileOffset, CF_TYPE_ANY);

		if (cfp == NULL)
			throw FFmpegException("Failed to open file.");

		m_ctx = FFmpegContext::createContext(cfp);
		auto ctx = m_ctx->ctx();

		AVCodec* audio_codec = nullptr;
		m_audioStreamIndex = av_find_best_stream(ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &audio_codec, 0);
		if (m_audioStreamIndex < 0) {
			throw FFmpegException("Failed to find audio stream in file.");
		}
		m_audioStream = ctx->streams[m_audioStreamIndex];

		int err;
#if LIBAVCODEC_VERSION_INT > AV_VERSION_INT(57, 24, 255)
		m_audioCodecCtx = avcodec_alloc_context3(audio_codec);

		// Copy codec parameters from input stream to output codec context
		err = avcodec_parameters_to_context(m_audioCodecCtx, m_audioStream->codecpar);
		if (err < 0) {
			char errorStr[512];
			av_strerror(err, errorStr, sizeof(errorStr));
			throw FFmpegException(errorStr);
		}
#else
		m_audioCodecCtx = m_audioStream->codec;
#endif

		err = avcodec_open2(m_audioCodecCtx, audio_codec, nullptr);
		if (err < 0) {
			char errorStr[512];
			av_strerror(err, errorStr, sizeof(errorStr));
			throw FFmpegException(errorStr);
		}

		m_baseAudioProps = getAudioProps(m_audioStream);

		m_audioProps = getAdjustedAudioProps(m_baseAudioProps);

		m_resampleCtx = getSWRContext(m_baseAudioProps, m_audioProps);

		m_al_format = openal_get_format(av_get_bytes_per_sample(m_audioProps.format) * 8, m_audioProps.channels);

		if (m_al_format == AL_INVALID_VALUE) {
			throw FFmpegException("Invalid audio format.");
		}
	} catch (const FFmpegException& e) {
		nprintf(("SOUND", "SOUND ==> Could not open wave file %s for streaming. Reason: %s\n", filename, e.what()));
		return false;
	}

	// Cue for streaming
	Cue();

	nprintf(("SOUND", "AUDIOSTR => Successfully opened: %s\n", filename));

	// If we are here it means that everything went fine
	return true;
}


bool WaveFile::Cue()
{
	auto err = av_seek_frame(m_ctx->ctx(), m_audioStreamIndex, 0, 0);
	avcodec_flush_buffers(m_audioCodecCtx);

	return err >= 0;
}

size_t WaveFile::handleDecodedFrame(AVFrame* av_frame, uint8_t* out_buffer, size_t buffer_size) {
	const auto sample_size = (av_get_bytes_per_sample(m_audioProps.format) * m_audioProps.channels);
	int dest_num_samples = static_cast<int>(buffer_size / sample_size);
	auto written = swr_convert(m_resampleCtx,
							   &out_buffer,
							   dest_num_samples,
							   (const uint8_t**) av_frame->data,
							   av_frame->nb_samples);


	return (size_t) (written * sample_size);
}

size_t WaveFile::getBufferedData(uint8_t* buffer, size_t buffer_size) {
	// First determine if there is buffered data
	auto buffered = swr_get_out_samples(m_resampleCtx, 0);

	if (buffered > 0) {
		const auto sample_size = (av_get_bytes_per_sample(m_audioProps.format) * m_audioProps.channels);
		int dest_num_samples = static_cast<int>(buffer_size / sample_size);
		auto written = swr_convert(m_resampleCtx, &buffer, dest_num_samples, nullptr, 0);

		auto advance = (size_t)(written * sample_size);
		Assertion(advance <= buffer_size,
				  "Buffer overrun!!! Decoding has written more data into the buffer than available!");

		return advance;
	}
	return 0;
}

int WaveFile::Read(uint8_t* pbDest, size_t cbSize)
{
	size_t buffer_pos = 0;
	buffer_pos += getBufferedData(pbDest, cbSize);
	if (buffer_pos == cbSize) {
		return static_cast<int>(buffer_pos);
	}

	FFmpegAudioReader reader(m_ctx->ctx(), m_audioCodecCtx, m_audioStreamIndex);

	while (reader.readFrame(m_decodeFrame)) {
		// Got a new frame
		auto advance = handleDecodedFrame(m_decodeFrame, pbDest + buffer_pos, cbSize - buffer_pos);

		if (advance == 0) {
			// No new data, return what we currently have
			return (int) buffer_pos;
		}

		buffer_pos += advance;
		Assertion(buffer_pos <= cbSize,
				  "Buffer overrun!!! Decoding has written more data into the buffer than available!");

		if (buffer_pos == cbSize) {
			return static_cast<int>(buffer_pos);
		}
	}

	// If we are here then the audio stream is finished, write any buffered data
	buffer_pos += getBufferedData(pbDest + buffer_pos, cbSize - buffer_pos);

	if (buffer_pos == 0) {
		// End of stream has been reached
		return -1;
	}

	return static_cast<int>(buffer_pos);
}

int WaveFile::getSampleByteSize() const {
	return av_get_bytes_per_sample(m_audioProps.format) * m_audioProps.channels;
}

int WaveFile::getSampleRate() const {
	return m_audioProps.sample_rate;
}
int WaveFile::getTotalSamples() const {
	auto duration = m_audioStream->time_base;
	duration.num *= (int)m_audioStream->duration;

	auto samples_r = av_mul_q(av_make_q(m_audioProps.sample_rate, 1), duration);

	// Compute the actual sample number and round the result up
	auto samples = 1 + ((samples_r.num - 1) / samples_r.den);

	return samples;
}

}