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authorMax Kellermann <max@duempel.org>2008-08-26 08:27:11 +0200
committerMax Kellermann <max@duempel.org>2008-08-26 08:27:11 +0200
commite99333167e152dcdd3b40ae812f920eb5023543a (patch)
tree3e9358e115aebac8e3bf601a62455a5c99e4f196 /src/inputPlugins/aac_plugin.c
parent5300f79ca98c55b367a612c93f0d7b2eef72ad23 (diff)
downloadmpd-e99333167e152dcdd3b40ae812f920eb5023543a.tar.gz
mpd-e99333167e152dcdd3b40ae812f920eb5023543a.tar.xz
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aac: support decoding AAC streams
Copy some code from aac_decode() to aac_stream_decode() and apply necessary changes to allow streaming audio data. Both functions might be merged later.
Diffstat (limited to '')
-rw-r--r--src/inputPlugins/aac_plugin.c139
1 files changed, 137 insertions, 2 deletions
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index a01d4ccbf..c168e9bef 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -300,6 +300,141 @@ static int getAacTotalTime(char *file)
return file_time;
}
+static int aac_stream_decode(struct decoder * mpd_decoder,
+ InputStream *inStream)
+{
+ float file_time;
+ float totalTime = 0;
+ faacDecHandle decoder;
+ faacDecFrameInfo frameInfo;
+ faacDecConfigurationPtr config;
+ long bread;
+ AudioFormat audio_format;
+ uint32_t sampleRate;
+ unsigned char channels;
+ unsigned int sampleCount;
+ char *sampleBuffer;
+ size_t sampleBufferLen;
+ mpd_uint16 bitRate = 0;
+ AacBuffer b;
+ int initialized = 0;
+
+ initAacBuffer(inStream, &b);
+
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+ config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+ config->dontUpSampleImplicitSBR = 0;
+#endif
+ faacDecSetConfiguration(decoder, config);
+
+ while (b.bytesIntoBuffer < FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS &&
+ !b.atEof &&
+ decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
+ fillAacBuffer(&b);
+ adts_find_frame(&b);
+ fillAacBuffer(&b);
+ my_usleep(10000);
+ }
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
+ &sampleRate, &channels);
+#else
+ bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+#endif
+ if (bread < 0) {
+ ERROR("Error not a AAC stream.\n");
+ faacDecClose(decoder);
+ if (b.buffer)
+ free(b.buffer);
+ return -1;
+ }
+
+ audio_format.bits = 16;
+
+ file_time = 0.0;
+
+ advanceAacBuffer(&b, bread);
+
+ while (1) {
+ fillAacBuffer(&b);
+ adts_find_frame(&b);
+ fillAacBuffer(&b);
+
+ if (b.bytesIntoBuffer == 0)
+ break;
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer,
+ b.bytesIntoBuffer);
+#else
+ sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer);
+#endif
+
+ if (frameInfo.error > 0) {
+ ERROR("error decoding AAC stream\n");
+ ERROR("faad2 error: %s\n",
+ faacDecGetErrorMessage(frameInfo.error));
+ break;
+ }
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+ sampleRate = frameInfo.samplerate;
+#endif
+
+ if (!initialized) {
+ audio_format.channels = frameInfo.channels;
+ audio_format.sampleRate = sampleRate;
+ decoder_initialized(mpd_decoder, &audio_format, totalTime);
+ initialized = 1;
+ }
+
+ advanceAacBuffer(&b, frameInfo.bytesconsumed);
+
+ sampleCount = (unsigned long)(frameInfo.samples);
+
+ if (sampleCount > 0) {
+ bitRate = frameInfo.bytesconsumed * 8.0 *
+ frameInfo.channels * sampleRate /
+ frameInfo.samples / 1000 + 0.5;
+ file_time +=
+ (float)(frameInfo.samples) / frameInfo.channels /
+ sampleRate;
+ }
+
+ sampleBufferLen = sampleCount * 2;
+
+ decoder_data(mpd_decoder, NULL, 0, sampleBuffer,
+ sampleBufferLen, file_time,
+ bitRate, NULL);
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ decoder_seek_error(mpd_decoder);
+ } else if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
+ break;
+ }
+
+ decoder_flush(mpd_decoder);
+
+ faacDecClose(decoder);
+ if (b.buffer)
+ free(b.buffer);
+
+ if (!initialized)
+ return -1;
+
+ if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
+ decoder_seek_error(mpd_decoder);
+ }
+
+ return 0;
+}
+
+
static int aac_decode(struct decoder * mpd_decoder, char *path)
{
float file_time;
@@ -461,10 +596,10 @@ struct decoder_plugin aacPlugin = {
NULL,
NULL,
NULL,
- NULL,
+ aac_stream_decode,
aac_decode,
aacTagDup,
- INPUT_PLUGIN_STREAM_FILE,
+ INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
aac_suffixes,
aac_mimeTypes
};