FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavcodec/roqaudioenc.c	Lines:	71	81	87.7 %
Date:	2020-10-23 17:01:47	Branches:	34	46	73.9 %


/*
 * RoQ audio encoder
 *
 * Copyright (c) 2005 Eric Lasota
 *    Based on RoQ specs (c)2001 Tim Ferguson
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "avcodec.h"
#include "bytestream.h"
#include "internal.h"
#include "mathops.h"

#define ROQ_FRAME_SIZE           735
#define ROQ_HEADER_SIZE   8

#define MAX_DPCM (127*127)


typedef struct ROQDPCMContext {
    short lastSample[2];
    int input_frames;
    int buffered_samples;
    int16_t *frame_buffer;
    int64_t first_pts;
} ROQDPCMContext;


static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
    ROQDPCMContext *context = avctx->priv_data;

    av_freep(&context->frame_buffer);

    return 0;
}

static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
{
    ROQDPCMContext *context = avctx->priv_data;

    if (avctx->channels > 2) {
        av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
        return AVERROR(EINVAL);
    }
    if (avctx->sample_rate != 22050) {
        av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
        return AVERROR(EINVAL);
    }

    avctx->frame_size = ROQ_FRAME_SIZE;
    avctx->bit_rate   = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
                        (22050 / ROQ_FRAME_SIZE) * 8;

    context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
                                      sizeof(*context->frame_buffer));
    if (!context->frame_buffer)
        return AVERROR(ENOMEM);

    context->lastSample[0] = context->lastSample[1] = 0;

    return 0;
}

static unsigned char dpcm_predict(short *previous, short current)
{
    int diff;
    int negative;
    int result;
    int predicted;

    diff = current - *previous;

    negative = diff<0;
    diff = FFABS(diff);

    if (diff >= MAX_DPCM)
        result = 127;
    else {
        result = ff_sqrt(diff);
        result += diff > result*result+result;
    }

    /* See if this overflows */
 retry:
    diff = result*result;
    if (negative)
        diff = -diff;
    predicted = *previous + diff;

    /* If it overflows, back off a step */
    if (predicted > 32767 || predicted < -32768) {
        result--;
        goto retry;
    }

    /* Add the sign bit */
    result |= negative << 7;   //if (negative) result |= 128;

    *previous = predicted;

    return result;
}

static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                                 const AVFrame *frame, int *got_packet_ptr)
{
    int i, stereo, data_size, ret;
    const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
    uint8_t *out;
    ROQDPCMContext *context = avctx->priv_data;

    stereo = (avctx->channels == 2);

    if (!in && context->input_frames >= 8)
        return 0;

    if (in && context->input_frames < 8) {
        memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
               in, avctx->frame_size * avctx->channels * sizeof(*in));
        context->buffered_samples += avctx->frame_size;
        if (context->input_frames == 0)
            context->first_pts = frame->pts;
        if (context->input_frames < 7) {
            context->input_frames++;
            return 0;
        }
    }
    if (context->input_frames < 8)
        in = context->frame_buffer;

    if (stereo) {
        context->lastSample[0] &= 0xFF00;
        context->lastSample[1] &= 0xFF00;
    }

    if (context->input_frames == 7)
        data_size = avctx->channels * context->buffered_samples;
    else
        data_size = avctx->channels * avctx->frame_size;

    if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size, 0)) < 0)
        return ret;
    out = avpkt->data;

    bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
    bytestream_put_byte(&out, 0x10);
    bytestream_put_le32(&out, data_size);

    if (stereo) {
        bytestream_put_byte(&out, (context->lastSample[1])>>8);
        bytestream_put_byte(&out, (context->lastSample[0])>>8);
    } else
        bytestream_put_le16(&out, context->lastSample[0]);

    /* Write the actual samples */
    for (i = 0; i < data_size; i++)
        *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);

    avpkt->pts      = context->input_frames <= 7 ? context->first_pts : frame->pts;
    avpkt->duration = data_size / avctx->channels;

    context->input_frames++;
    if (!in)
        context->input_frames = FFMAX(context->input_frames, 8);

    *got_packet_ptr = 1;
    return 0;
}

AVCodec ff_roq_dpcm_encoder = {
    .name           = "roq_dpcm",
    .long_name      = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_CODEC_ID_ROQ_DPCM,
    .priv_data_size = sizeof(ROQDPCMContext),
    .init           = roq_dpcm_encode_init,
    .encode2        = roq_dpcm_encode_frame,
    .close          = roq_dpcm_encode_close,
    .capabilities   = AV_CODEC_CAP_DELAY,
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                     AV_SAMPLE_FMT_NONE },
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* RoQ audio encoder
3			*
4			* Copyright (c) 2005 Eric Lasota
5			* Based on RoQ specs (c)2001 Tim Ferguson
6			*
7			* This file is part of FFmpeg.
8			*
9			* FFmpeg is free software; you can redistribute it and/or
10			* modify it under the terms of the GNU Lesser General Public
11			* License as published by the Free Software Foundation; either
12			* version 2.1 of the License, or (at your option) any later version.
13			*
14			* FFmpeg is distributed in the hope that it will be useful,
15			* but WITHOUT ANY WARRANTY; without even the implied warranty of
16			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17			* Lesser General Public License for more details.
18			*
19			* You should have received a copy of the GNU Lesser General Public
20			* License along with FFmpeg; if not, write to the Free Software
21			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22			*/
23
24			#include "avcodec.h"
25			#include "bytestream.h"
26			#include "internal.h"
27			#include "mathops.h"
28
29			#define ROQ_FRAME_SIZE 735
30			#define ROQ_HEADER_SIZE 8
31
32			#define MAX_DPCM (127*127)
33
34
35			typedef struct ROQDPCMContext {
36			short lastSample[2];
37			int input_frames;
38			int buffered_samples;
39			int16_t *frame_buffer;
40			int64_t first_pts;
41			} ROQDPCMContext;
42
43
44		1	static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
45			{
46		1	ROQDPCMContext *context = avctx->priv_data;
47
48		1	av_freep(&context->frame_buffer);
49
50		1	return 0;
51			}
52
53		1	static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
54			{
55		1	ROQDPCMContext *context = avctx->priv_data;
56
57	✗✓	1	if (avctx->channels > 2) {
58			av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
59			return AVERROR(EINVAL);
60			}
61	✗✓	1	if (avctx->sample_rate != 22050) {
62			av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
63			return AVERROR(EINVAL);
64			}
65
66		1	avctx->frame_size = ROQ_FRAME_SIZE;
67		1	avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
68		1	(22050 / ROQ_FRAME_SIZE) * 8;
69
70		1	context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
71			sizeof(*context->frame_buffer));
72	✗✓	1	if (!context->frame_buffer)
73			return AVERROR(ENOMEM);
74
75		1	context->lastSample[0] = context->lastSample[1] = 0;
76
77		1	return 0;
78			}
79
80		264600	static unsigned char dpcm_predict(short *previous, short current)
81			{
82			int diff;
83			int negative;
84			int result;
85			int predicted;
86
87		264600	diff = current - *previous;
88
89		264600	negative = diff<0;
90		264600	diff = FFABS(diff);
91
92	✓✓	264600	if (diff >= MAX_DPCM)
93		10896	result = 127;
94			else {
95		253704	result = ff_sqrt(diff);
96		253704	result += diff > result*result+result;
97			}
98
99			/* See if this overflows */
100		264600	retry:
101		264600	diff = result*result;
102	✓✓	264600	if (negative)
103		131603	diff = -diff;
104		264600	predicted = *previous + diff;
105
106			/* If it overflows, back off a step */
107	✓✗✗✓	264600	if (predicted > 32767 \|\| predicted < -32768) {
108			result--;
109			goto retry;
110			}
111
112			/* Add the sign bit */
113		264600	result \|= negative << 7; //if (negative) result \|= 128;
114
115		264600	*previous = predicted;
116
117		264600	return result;
118			}
119
120		181	static int roq_dpcm_encode_frame(AVCodecContext avctx, AVPacket avpkt,
121			const AVFrame frame, int got_packet_ptr)
122			{
123			int i, stereo, data_size, ret;
124	✓✓	181	const int16_t in = frame ? (const int16_t )frame->data[0] : NULL;
125			uint8_t *out;
126		181	ROQDPCMContext *context = avctx->priv_data;
127
128		181	stereo = (avctx->channels == 2);
129
130	✓✓✓✗	181	if (!in && context->input_frames >= 8)
131		1	return 0;
132
133	✓✗✓✓	180	if (in && context->input_frames < 8) {
134		8	memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
135		8	in, avctx->frame_size * avctx->channels * sizeof(*in));
136		8	context->buffered_samples += avctx->frame_size;
137	✓✓	8	if (context->input_frames == 0)
138		1	context->first_pts = frame->pts;
139	✓✓	8	if (context->input_frames < 7) {
140		7	context->input_frames++;
141		7	return 0;
142			}
143			}
144	✓✓	173	if (context->input_frames < 8)
145		1	in = context->frame_buffer;
146
147	✓✗	173	if (stereo) {
148		173	context->lastSample[0] &= 0xFF00;
149		173	context->lastSample[1] &= 0xFF00;
150			}
151
152	✓✓	173	if (context->input_frames == 7)
153		1	data_size = avctx->channels * context->buffered_samples;
154			else
155		172	data_size = avctx->channels * avctx->frame_size;
156
157	✗✓	173	if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size, 0)) < 0)
158			return ret;
159		173	out = avpkt->data;
160
161	✓✗	173	bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
162		173	bytestream_put_byte(&out, 0x10);
163		173	bytestream_put_le32(&out, data_size);
164
165	✓✗	173	if (stereo) {
166		173	bytestream_put_byte(&out, (context->lastSample[1])>>8);
167		173	bytestream_put_byte(&out, (context->lastSample[0])>>8);
168			} else
169			bytestream_put_le16(&out, context->lastSample[0]);
170
171			/* Write the actual samples */
172	✓✓	264773	for (i = 0; i < data_size; i++)
173		264600	out++ = dpcm_predict(&context->lastSample[i & 1], in++);
174
175	✓✓	173	avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
176		173	avpkt->duration = data_size / avctx->channels;
177
178		173	context->input_frames++;
179	✗✓	173	if (!in)
180			context->input_frames = FFMAX(context->input_frames, 8);
181
182		173	*got_packet_ptr = 1;
183		173	return 0;
184			}
185
186			AVCodec ff_roq_dpcm_encoder = {
187			.name = "roq_dpcm",
188			.long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
189			.type = AVMEDIA_TYPE_AUDIO,
190			.id = AV_CODEC_ID_ROQ_DPCM,
191			.priv_data_size = sizeof(ROQDPCMContext),
192			.init = roq_dpcm_encode_init,
193			.encode2 = roq_dpcm_encode_frame,
194			.close = roq_dpcm_encode_close,
195			.capabilities = AV_CODEC_CAP_DELAY,
196			.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
197			AV_SAMPLE_FMT_NONE },
198			};