decoder: use audio_format_init_checked()

Let the audio_check library verify the audio format in all (relevant,
i.e. non-hardcoded) plugins.

Makefile.am

@@ -789,6 +789,7 @@ test_run_decoder_SOURCES = test/run_decoder.c \
 	src/replay_gain.c \
 	src/uri.c \
 	src/fd_util.c \
+	src/audio_check.c \
 	$(ARCHIVE_SRC) \
 	$(INPUT_SRC) \
 	$(TAG_SRC) \
@@ -809,6 +810,7 @@ test_read_tags_SOURCES = test/read_tags.c \
 	src/replay_gain.c \
 	src/uri.c \
 	src/fd_util.c \
+	src/audio_check.c \
 	$(ARCHIVE_SRC) \
 	$(INPUT_SRC) \
 	$(TAG_SRC) \

src/decoder/_flac_common.c

@@ -25,6 +25,7 @@
 #include "_flac_common.h"
 #include "flac_metadata.h"
 #include "flac_pcm.h"
+#include "audio_check.h"
 
 #include <glib.h>
 
@@ -63,20 +64,19 @@ bool
 flac_data_get_audio_format(struct flac_data *data,
 			   struct audio_format *audio_format)
 {
+	GError *error = NULL;
+
 	if (!data->have_stream_info) {
 		g_warning("no STREAMINFO packet found");
 		return false;
 	}
 
-	audio_format_init(audio_format, data->stream_info.sample_rate,
-			  data->stream_info.bits_per_sample,
-			  data->stream_info.channels);
-
-	if (!audio_format_valid(audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format->sample_rate,
-			  audio_format->bits,
-			  audio_format->channels);
+	if (!audio_format_init_checked(audio_format,
+				       data->stream_info.sample_rate,
+				       data->stream_info.bits_per_sample,
+				       data->stream_info.channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		return false;
 	}
 

src/decoder/audiofile_plugin.c

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "decoder_api.h"
+#include "audio_check.h"
 
 #include <audiofile.h>
 #include <af_vfs.h>
@@ -103,6 +104,7 @@ setup_virtual_fops(struct input_stream *stream)
 static void
 audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 {
+	GError *error = NULL;
 	AFvirtualfile *vf;
 	int fs, frame_count;
 	AFfilehandle af_fp;
@@ -138,13 +140,13 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 	                         AF_SAMPFMT_TWOSCOMP, bits);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
 
-	audio_format_init(&audio_format, afGetRate(af_fp, AF_DEFAULT_TRACK),
-			  bits, afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK)); 
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate, audio_format.bits,
-			  audio_format.channels);
+	if (!audio_format_init_checked(&audio_format,
+				       afGetRate(af_fp, AF_DEFAULT_TRACK),
+				       bits,
+				       afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK),
+				       &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		afCloseFile(af_fp);
 		return;
 	}

src/decoder/faad_plugin.c

@@ -20,6 +20,7 @@
 #include "config.h"
 #include "decoder_api.h"
 #include "decoder_buffer.h"
+#include "audio_check.h"
 
 #define AAC_MAX_CHANNELS	6
 
@@ -36,6 +37,15 @@ static const unsigned adts_sample_rates[] =
 	16000, 12000, 11025, 8000, 7350, 0, 0, 0
 };
 
+/**
+ * The GLib quark used for errors reported by this plugin.
+ */
+static inline GQuark
+faad_decoder_quark(void)
+{
+	return g_quark_from_static_string("faad");
+}
+
 /**
  * Check whether the buffer head is an AAC frame, and return the frame
  * length.  Returns 0 if it is not a frame.
@@ -232,7 +242,7 @@ faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is)
  */
 static bool
 faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
-		  struct audio_format *audio_format)
+		  struct audio_format *audio_format, GError **error_r)
 {
 	union {
 		/* deconst hack for libfaad */
@@ -247,28 +257,33 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
 	/* neaacdec.h declares all arguments as "unsigned long", but
 	   internally expects uint32_t pointers.  To avoid gcc
 	   warnings, use this workaround. */
-	unsigned long *sample_rate_r = (unsigned long *)(void *)&sample_rate;
+	unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate;
 #else
-	uint32_t *sample_rate_r = &sample_rate;
+	uint32_t *sample_rate_p = &sample_rate;
 #endif
 
 	u.in = decoder_buffer_read(buffer, &length);
-	if (u.in == NULL)
+	if (u.in == NULL) {
+		g_set_error(error_r, faad_decoder_quark(), 0,
+			    "Empty file");
 		return false;
+	}
 
 	nbytes = faacDecInit(decoder, u.out,
 #ifdef HAVE_FAAD_BUFLEN_FUNCS
 			     length,
 #endif
-			     sample_rate_r, &channels);
-	if (nbytes < 0)
+			     sample_rate_p, &channels);
+	if (nbytes < 0) {
+		g_set_error(error_r, faad_decoder_quark(), 0,
+			    "Not an AAC stream");
 		return false;
+	}
 
 	decoder_buffer_consume(buffer, nbytes);
 
-	audio_format_init(audio_format, sample_rate, 16, channels);
-
-	return true;
+	return audio_format_init_checked(audio_format, sample_rate,
+					 16, channels, error_r);
 }
 
 /**
@@ -334,8 +349,8 @@ faad_get_file_time_float(const char *file)
 
 		decoder_buffer_fill(buffer);
 
-		ret = faad_decoder_init(decoder, buffer, &audio_format);
-		if (ret && audio_format_valid(&audio_format))
+		ret = faad_decoder_init(decoder, buffer, &audio_format, NULL);
+		if (ret)
 			length = 0;
 
 		faacDecClose(decoder);
@@ -367,6 +382,7 @@ faad_get_file_time(const char *file)
 static void
 faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
 {
+	GError *error = NULL;
 	float file_time;
 	float total_time = 0;
 	faacDecHandle decoder;
@@ -404,15 +420,10 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
 
 	/* initialize it */
 
-	ret = faad_decoder_init(decoder, buffer, &audio_format);
+	ret = faad_decoder_init(decoder, buffer, &audio_format, &error);
 	if (!ret) {
-		g_warning("Error not a AAC stream.\n");
-		faacDecClose(decoder);
-		return;
-	}
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("invalid audio format\n");
+		g_warning("%s", error->message);
+		g_error_free(error);
 		faacDecClose(decoder);
 		return;
 	}

src/decoder/ffmpeg_plugin.c

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "decoder_api.h"
+#include "audio_check.h"
 
 #include <glib.h>
 
@@ -260,6 +261,7 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 static bool
 ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 {
+	GError *error = NULL;
 	struct decoder *decoder = ctx->decoder;
 	AVCodecContext *codec_context = ctx->codec_context;
 	AVFormatContext *format_context = ctx->format_context;
@@ -281,13 +283,11 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 	/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
 	bits = (uint8_t) 16;
 #endif
-	audio_format_init(&audio_format, codec_context->sample_rate, bits,
-			  codec_context->channels);
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate, audio_format.bits,
-			  audio_format.channels);
+	if (!audio_format_init_checked(&audio_format,
+				       codec_context->sample_rate, bits,
+				       codec_context->channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		return false;
 	}
 

src/decoder/mad_plugin.c

@@ -21,6 +21,7 @@
 #include "decoder_api.h"
 #include "conf.h"
 #include "tag_id3.h"
+#include "audio_check.h"
 
 #include <assert.h>
 #include <unistd.h>
@@ -1174,6 +1175,7 @@ static void
 mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 {
 	struct mp3_data data;
+	GError *error = NULL;
 	struct tag *tag = NULL;
 	struct replay_gain_info *replay_gain_info = NULL;
 	struct audio_format audio_format;
@@ -1185,8 +1187,20 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 		return;
 	}
 
-	audio_format_init(&audio_format, data.frame.header.samplerate, 24,
-			  MAD_NCHANNELS(&data.frame.header));
+	if (!audio_format_init_checked(&audio_format,
+				       data.frame.header.samplerate, 24,
+				       MAD_NCHANNELS(&data.frame.header),
+				       &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+
+		if (tag != NULL)
+			tag_free(tag);
+		if (replay_gain_info != NULL)
+			replay_gain_info_free(replay_gain_info);
+		mp3_data_finish(&data);
+		return;
+	}
 
 	decoder_initialized(decoder, &audio_format,
 			    data.input_stream->seekable, data.total_time);

src/decoder/mikmod_plugin.c

@@ -22,6 +22,7 @@
 
 #include <glib.h>
 #include <mikmod.h>
+#include <assert.h>
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "mikmod"
@@ -177,6 +178,7 @@ mod_decode(struct decoder *decoder, const char *path)
 	}
 
 	audio_format_init(&audio_format, 44100, 16, 2);
+	assert(audio_format_valid(&audio_format));
 
 	secPerByte =
 	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *

src/decoder/modplug_plugin.c

@@ -22,6 +22,7 @@
 
 #include <glib.h>
 #include <modplug.h>
+#include <assert.h>
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "modplug"
@@ -123,6 +124,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
 	}
 
 	audio_format_init(&audio_format, 44100, 16, 2);
+	assert(audio_format_valid(&audio_format));
 
 	sec_perbyte =
 	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *

src/decoder/mp4ff_plugin.c

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "decoder_api.h"
+#include "audio_check.h"
 
 #include <glib.h>
 
@@ -110,6 +111,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
 	int track;
 	uint32_t sample_rate;
 	unsigned char channels;
+	GError *error = NULL;
 
 	decoder = faacDecOpen();
 
@@ -130,18 +132,16 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
 		return NULL;
 	}
 
-	*track_r = track;
-	audio_format_init(audio_format, sample_rate, 16, channels);
-
-	if (!audio_format_valid(audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format->sample_rate,
-			  audio_format->bits,
-			  audio_format->channels);
+	if (!audio_format_init_checked(audio_format, sample_rate, 16, channels,
+				       &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		faacDecClose(decoder);
 		return NULL;
 	}
 
+	*track_r = track;
+
 	return decoder;
 }
 

src/decoder/mpcdec_plugin.c

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "decoder_api.h"
+#include "audio_check.h"
 
 #ifdef MPC_IS_OLD_API
 #include <mpcdec/mpcdec.h>
@@ -27,6 +28,7 @@
 #endif
 
 #include <glib.h>
+#include <assert.h>
 #include <unistd.h>
 
 #undef G_LOG_DOMAIN
@@ -140,6 +142,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 #endif
 	mpc_reader reader;
 	mpc_streaminfo info;
+	GError *error = NULL;
 	struct audio_format audio_format;
 
 	struct mpc_decoder_data data;
@@ -193,16 +196,13 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 	mpc_demux_get_info(demux, &info);
 #endif
 
-	audio_format_init(&audio_format, info.sample_freq, 24, info.channels);
-
-	if (!audio_format_valid(&audio_format)) {
+	if (!audio_format_init_checked(&audio_format, info.sample_freq, 16,
+				       info.channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 #ifndef MPC_IS_OLD_API
 		mpc_demux_exit(demux);
 #endif
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate,
-			  audio_format.bits,
-			  audio_format.channels);
 		return;
 	}
 

src/decoder/mpg123_decoder_plugin.c

@@ -19,6 +19,7 @@
 
 #include "config.h" /* must be first for large file support */
 #include "decoder_api.h"
+#include "audio_check.h"
 
 #include <glib.h>
 
@@ -54,6 +55,7 @@ static bool
 mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
 		struct audio_format *audio_format)
 {
+	GError *gerror = NULL;
 	char *path_dup;
 	int error;
 	int channels, encoding;
@@ -85,9 +87,10 @@ mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
 		return false;
 	}
 
-	audio_format_init(audio_format, rate, 16, channels);
-	if (!audio_format_valid(audio_format)) {
-		g_warning("invalid audio format");
+	if (!audio_format_init_checked(audio_format, rate, 16,
+				       channels, &gerror)) {
+		g_warning("%s", gerror->message);
+		g_error_free(gerror);
 		return false;
 	}
 

src/decoder/sidplay_plugin.cxx

@@ -278,6 +278,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
 
 	struct audio_format audio_format;
 	audio_format_init(&audio_format, 48000, 16, 2);
+	assert(audio_format_valid(&audio_format));
 
 	decoder_initialized(decoder, &audio_format, true, (float)song_len);
 

src/decoder/sndfile_decoder_plugin.c

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "decoder_api.h"
+#include "audio_check.h"
 
 #include <sndfile.h>
 
@@ -109,6 +110,7 @@ time_to_frame(float t, const struct audio_format *audio_format)
 static void
 sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
 {
+	GError *error = NULL;
 	SNDFILE *sf;
 	SF_INFO info;
 	struct audio_format audio_format;
@@ -128,10 +130,10 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
 	/* for now, always read 32 bit samples.  Later, we could lower
 	   MPD's CPU usage by reading 16 bit samples with
 	   sf_readf_short() on low-quality source files. */
-	audio_format_init(&audio_format, info.samplerate, 32, info.channels);
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("invalid audio format");
+	if (!audio_format_init_checked(&audio_format, info.samplerate, 32,
+				       info.channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		return;
 	}
 

src/decoder/vorbis_plugin.c

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "_ogg_common.h"
+#include "audio_check.h"
 #include "uri.h"
 
 #ifndef HAVE_TREMOR
@@ -264,6 +265,7 @@ static void
 vorbis_stream_decode(struct decoder *decoder,
 		     struct input_stream *input_stream)
 {
+	GError *error = NULL;
 	OggVorbis_File vf;
 	ov_callbacks callbacks;
 	struct vorbis_decoder_data data;
@@ -309,13 +311,10 @@ vorbis_stream_decode(struct decoder *decoder,
 		return;
 	}
 
-	audio_format_init(&audio_format, vi->rate, 16, vi->channels);
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate,
-			  audio_format.bits,
-			  audio_format.channels);
+	if (!audio_format_init_checked(&audio_format, vi->rate, 16,
+				       vi->channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		return;
 	}
 

src/decoder/wavpack_plugin.c

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "decoder_api.h"
+#include "audio_check.h"
 #include "path.h"
 #include "utils.h"
 
@@ -130,6 +131,8 @@ static void
 wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 	       struct replay_gain_info *replay_gain_info)
 {
+	GError *error = NULL;
+	unsigned bits;
 	struct audio_format audio_format;
 	format_samples_t format_samples;
 	char chunk[CHUNK_SIZE];
@@ -138,24 +141,22 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 	int bytes_per_sample, output_sample_size;
 	int position;
 
-	audio_format_init(&audio_format, WavpackGetSampleRate(wpc),
-			  WavpackGetBitsPerSample(wpc), 
-			  WavpackGetNumChannels(wpc));
+	bits = WavpackGetBitsPerSample(wpc);
 
 	/* round bitwidth to 8-bit units */
-	audio_format.bits = (audio_format.bits + 7) & (~7);
+	bits = (bits + 7) & (~7);
 	/* MPD handles max 32-bit samples */
-	if (audio_format.bits > 32)
-		audio_format.bits = 32;
+	if (bits > 32)
+		bits = 32;
 
 	if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
-		audio_format.bits = 24;
+		bits = 24;
 
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate,
-			  audio_format.bits,
-			  audio_format.channels);
+	if (!audio_format_init_checked(&audio_format,
+				       WavpackGetSampleRate(wpc), bits,
+				       WavpackGetNumChannels(wpc), &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		return;
 	}