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.
---
src/inputPlugins/aac_plugin.c | 139 ++++++++++++++++++++++++++++++++++++++++-
1 files changed, 137 insertions(+), 2 deletions(-)
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index a01d4cc..c168e9b 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
};
-------------------------------------------------------------------------
This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
Musicpd-dev-team mailing list
Musicpd-dev-team@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/musicpd-dev-team
