Update Myth source base to work with new FFmpeg following 99b0c9d

mythtv/libs/libmyth/audio/audiooutputbase.cpp

@@ -47,7 +47,7 @@ const char *AudioOutputBase::quality_string(int q)
 AudioOutputBase::AudioOutputBase(const AudioSettings &settings) :
     MThread("AudioOutputBase"),
     // protected
-    channels(-1),               codec(CODEC_ID_NONE),
+    channels(-1),               codec(AV_CODEC_ID_NONE),
     bytes_per_frame(0),         output_bytes_per_frame(0),
     format(FORMAT_NONE),        output_format(FORMAT_NONE),
     samplerate(-1),             effdsp(0),
@@ -251,10 +251,10 @@ bool AudioOutputBase::CanPassthrough(int samplerate, int channels,
 
     switch(codec)
     {
-        case CODEC_ID_AC3:
+        case AV_CODEC_ID_AC3:
             arg = FEATURE_AC3;
             break;
-        case CODEC_ID_DTS:
+        case AV_CODEC_ID_DTS:
             switch(profile)
             {
                 case FF_PROFILE_DTS:
@@ -270,10 +270,10 @@ bool AudioOutputBase::CanPassthrough(int samplerate, int channels,
                     break;
             }
             break;
-        case CODEC_ID_EAC3:
+        case AV_CODEC_ID_EAC3:
             arg = FEATURE_EAC3;
             break;
-        case CODEC_ID_TRUEHD:
+        case AV_CODEC_ID_TRUEHD:
             arg = FEATURE_TRUEHD;
             break;
     }
@@ -429,7 +429,7 @@ bool AudioOutputBase::CanUpmix(void)
 bool AudioOutputBase::SetupPassthrough(int codec, int codec_profile,
                                        int &samplerate_tmp, int &channels_tmp)
 {
-    if (codec == CODEC_ID_DTS &&
+    if (codec == AV_CODEC_ID_DTS &&
         !output_settingsdigital->canFeature(FEATURE_DTSHD))
     {
         // We do not support DTS-HD bitstream so force extraction of the
@@ -448,7 +448,7 @@ bool AudioOutputBase::SetupPassthrough(int codec, int codec_profile,
     }
 
     m_spdifenc = new SPDIFEncoder("spdif", codec);
-    if (m_spdifenc->Succeeded() && codec == CODEC_ID_DTS)
+    if (m_spdifenc->Succeeded() && codec == AV_CODEC_ID_DTS)
     {
         switch(codec_profile)
         {
@@ -553,7 +553,7 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings)
         /* Might we reencode a bitstream that's been decoded for timestretch?
            If the device doesn't support the number of channels - see below */
         if (output_settingsdigital->canFeature(FEATURE_AC3) &&
-            (settings.codec == CODEC_ID_AC3 || settings.codec == CODEC_ID_DTS))
+            (settings.codec == AV_CODEC_ID_AC3 || settings.codec == AV_CODEC_ID_DTS))
         {
             lreenc = true;
         }
@@ -734,7 +734,7 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings)
                 .arg(samplerate).arg(configured_channels));
 
         encoder = new AudioOutputDigitalEncoder();
-        if (!encoder->Init(CODEC_ID_AC3, 448000, samplerate,
+        if (!encoder->Init(AV_CODEC_ID_AC3, 448000, samplerate,
                            configured_channels))
         {
             Error("AC-3 encoder initialization failed");

mythtv/libs/libmyth/audio/audiooutputdigitalencoder.cpp

@@ -154,7 +154,7 @@ bool AudioOutputDigitalEncoder::Init(
         delete m_spdifenc;
     }
 
-    m_spdifenc = new SPDIFEncoder("spdif", CODEC_ID_AC3);
+    m_spdifenc = new SPDIFEncoder("spdif", AV_CODEC_ID_AC3);
     if (!m_spdifenc->Succeeded())
     {
         Dispose();
@@ -235,7 +235,7 @@ size_t AudioOutputDigitalEncoder::Encode(void *buf, int len, AudioFormat format)
 
         if (!m_spdifenc)
         {
-            m_spdifenc = new SPDIFEncoder("spdif", CODEC_ID_AC3);
+            m_spdifenc = new SPDIFEncoder("spdif", AV_CODEC_ID_AC3);
         }
         m_spdifenc->WriteFrame((uint8_t *)m_encodebuffer, outsize);
         // Check if output buffer is big enough

mythtv/libs/libmyth/audio/audiooutputsettings.cpp

@@ -449,14 +449,14 @@ QString AudioOutputSettings::GetPassthroughParams(int codec, int codec_profile,
 
     switch (codec)
     {
-        case CODEC_ID_AC3:
+        case AV_CODEC_ID_AC3:
             log = "AC3";
             break;
-        case CODEC_ID_EAC3:
+        case AV_CODEC_ID_EAC3:
             samplerate = samplerate * 4;
             log = "Dolby Digital Plus (E-AC3)";
             break;
-        case CODEC_ID_DTS:
+        case AV_CODEC_ID_DTS:
             switch(codec_profile)
             {
                 case FF_PROFILE_DTS_ES:
@@ -487,7 +487,7 @@ QString AudioOutputSettings::GetPassthroughParams(int codec, int codec_profile,
                     break;
             }
             break;
-        case CODEC_ID_TRUEHD:
+        case AV_CODEC_ID_TRUEHD:
             channels = 8;
             log = "TrueHD";
             switch(samplerate)

mythtv/libs/libmyth/audio/audiooutpututil.cpp

@@ -1,11 +1,18 @@
 #include <math.h>
-
-#include "mythconfig.h"
-#include "audiooutpututil.h"
 #include <sys/types.h>
 #include <inttypes.h>
 #include "bswap.h"
 
+#include "mythconfig.h"
+#include "mythlogging.h"
+#include "audiooutpututil.h"
+
+extern "C" {
+#include "libavcodec/avcodec.h"
+#include "libswresample/swresample.h"
+#include "pink.h"
+}
+
 #define LOC QString("AOUtil: ")
 
 #if ARCH_X86
@@ -675,4 +682,84 @@ char *AudioOutputUtil::GeneratePinkFrames(char *frames, int channels,
     }    
     return frames;
 }
-
+
+/**
+ * DecodeAudio
+ * Decode an audio packet, and compact it if data is planar
+ * Return negative error code if an error occurred during decoding
+ * or the number of bytes consumed from the input AVPacket
+ * data_size contains the size of decoded data copied into buffer
+ */
+int AudioOutputUtil::DecodeAudio(AVCodecContext *ctx,
+                                 uint8_t *buffer, int &data_size,
+                                 AVPacket *pkt)
+{
+    AVFrame frame;
+    int got_frame = 0;
+    int ret, ret2;
+
+    data_size = 0;
+    avcodec_get_frame_defaults(&frame);
+    ret = avcodec_decode_audio4(ctx, &frame, &got_frame, pkt);
+    if (ret < 0 || !got_frame)
+    {
+        LOG(VB_AUDIO, LOG_ERR, LOC +
+            QString("audio decode error: %1 (%2)")
+            .arg(av_err2str(ret)).arg(got_frame));
+        return ret;
+    }
+
+    AVSampleFormat format = (AVSampleFormat)frame.format;
+    if (!av_sample_fmt_is_planar(format))
+    {
+            // data is already compacted... simply copy it
+        data_size = av_samples_get_buffer_size(NULL, ctx->channels,
+                                               frame.nb_samples,
+                                               format, 1);
+        memcpy(buffer, frame.extended_data[0], data_size);
+        return ret;
+    }
+
+        // Need to find a valid channels layout, as not all codecs provide one
+    int64_t channel_layout =
+        frame.channel_layout && frame.channels == av_get_channel_layout_nb_channels(frame.channel_layout) ?
+        frame.channel_layout : av_get_default_channel_layout(frame.channels);
+    SwrContext *swr = swr_alloc_set_opts(NULL,
+                                         channel_layout,
+                                         av_get_packed_sample_fmt(format),
+                                         frame.sample_rate,
+                                         channel_layout,
+                                         format,
+                                         frame.sample_rate,
+                                         0, NULL);
+    if (!swr)
+    {
+        LOG(VB_AUDIO, LOG_ERR, LOC + "error allocating resampler context");
+        return AVERROR(ENOMEM);
+    }
+        /* initialize the resampling context */
+    ret2 = swr_init(swr);
+    if (ret2 < 0)
+    {
+        LOG(VB_AUDIO, LOG_ERR, LOC +
+            QString("error initializing resampler context (%1)")
+            .arg(av_err2str(ret2)));
+        swr_free(&swr);
+        return ret2;
+    }
+
+    uint8_t *out[] = {buffer};
+    ret2 = swr_convert(swr, out, frame.nb_samples,
+                       (const uint8_t **)frame.extended_data, frame.nb_samples);
+    swr_free(&swr);
+    if (ret2 < 0)
+    {
+        LOG(VB_AUDIO, LOG_ERR, LOC +
+            QString("error converting audio from planar format (%1)")
+            .arg(av_err2str(ret2)));
+        return ret2;
+    }
+    data_size = ret2 * frame.channels * av_get_bytes_per_sample(format);
+
+    return ret;
+}

mythtv/libs/libmyth/audio/audiooutpututil.h

@@ -2,9 +2,6 @@
 #define AUDIOOUTPUTUTIL_H_
 
 #include "audiooutputsettings.h"
-extern "C" {
-#include "pink.h"
-}
 
 class MPUBLIC AudioOutputUtil
 {
@@ -19,6 +16,9 @@ class MPUBLIC AudioOutputUtil
                             void *buffer, int bytes);
     static char *GeneratePinkFrames(char *frames, int channels,
                                     int channel, int count, int bits = 16);
+    static int DecodeAudio(AVCodecContext *ctx,
+                           uint8_t *buffer, int &data_size,
+                           AVPacket *pkt);
 };
 
 #endif

mythtv/libs/libmyth/libmyth.pro

@@ -87,20 +87,22 @@ LIBS += -L../libmythservicecontracts -lmythservicecontracts-$${LIBVERSION}
 LIBS += -L../../external/FFmpeg/libavcodec -lmythavcodec
 LIBS += -L../../external/FFmpeg/libavutil  -lmythavutil
 LIBS += -L../../external/FFmpeg/libavformat  -lmythavformat
+LIBS += -L../../external/FFmpeg/libswresample -lmythswresample
 
 POST_TARGETDEPS += ../libmythsamplerate/libmythsamplerate-$${MYTH_LIB_EXT}
 POST_TARGETDEPS += ../libmythsoundtouch/libmythsoundtouch-$${MYTH_LIB_EXT}
 POST_TARGETDEPS += ../libmythfreesurround/libmythfreesurround-$${MYTH_LIB_EXT}
 POST_TARGETDEPS += ../../external/FFmpeg/libavcodec/$$avLibName(avcodec)
 POST_TARGETDEPS += ../../external/FFmpeg/libavutil/$$avLibName(avutil)
+POST_TARGETDEPS += ../../external/FFmpeg/libswresample/$$avLibName(swresample)
 
 # Install headers so that plugins can compile independently
 inc.path = $${PREFIX}/include/mythtv/
 inc.files  = dialogbox.h mythcontext.h
 inc.files += mythwidgets.h remotefile.h oldsettings.h volumecontrol.h
 inc.files += settings.h uitypes.h mythdialogs.h
 inc.files += audio/audiooutput.h audio/audiosettings.h
-inc.files += audio/audiooutputsettings.h
+inc.files += audio/audiooutputsettings.h audio/audiooutpututil.h
 inc.files += audio/volumebase.h audio/eldutils.h
 inc.files += inetcomms.h mythwizard.h schemawizard.h
 inc.files += mythmediamonitor.h

mythtv/libs/libmythtv/AirPlay/mythraopconnection.cpp

@@ -9,6 +9,7 @@
 #include "serverpool.h"
 
 #include "audiooutput.h"
+#include "audiooutpututil.h"
 
 #include "mythraopdevice.h"
 #include "mythraopconnection.h"
@@ -615,33 +616,28 @@ uint32_t MythRAOPConnection::decodeAudioPacket(uint8_t type,
     tmp_pkt.size = len;
 
     uint32_t frames_added = 0;
+    uint8_t *samples = (uint8_t *)av_mallocz(AVCODEC_MAX_AUDIO_FRAME_SIZE);
     while (tmp_pkt.size > 0)
     {
-        uint8_t *samples = (uint8_t *)av_mallocz(AVCODEC_MAX_AUDIO_FRAME_SIZE);
-        AVFrame frame;
-        int got_frame = 0;
-
-        int ret = avcodec_decode_audio4(ctx, &frame, &got_frame, &tmp_pkt);
-
+        int data_size;
+        int ret = AudioOutputUtil::DecodeAudio(ctx, samples,
+                                               data_size, &tmp_pkt);
         if (ret < 0)
         {
             av_free(samples);
             return -1;
         }
 
-        if (got_frame)
+        if (data_size)
         {
-            // ALAC codec isn't planar
-            int data_size = av_samples_get_buffer_size(NULL, ctx->channels,
-                                                       frame.nb_samples,
-                                                       ctx->sample_fmt, 1);
-            memcpy(samples, frame.extended_data[0], data_size);
+            int num_samples = data_size /
+	      (ctx->channels * av_get_bytes_per_sample(ctx->sample_fmt));
 
-            frames_added += frame.nb_samples;
+            frames_added += num_samples;
             AudioData block;
             block.data    = samples;
             block.length  = data_size;
-            block.frames  = frame.nb_samples;
+            block.frames  = num_samples;
             dest->append(block);
         }
         tmp_pkt.data += ret;